Mercurial > almixer_isolated
annotate ALmixer.c @ 12:bfe90b4f3d87
Bug fixes to FadeIn.
Documentation fixes
Minor API changes w.r.t. return types. (ALboolean instead of ALint)
| author | Eric Wing <ewing . public |-at-| gmail . com> |
|---|---|
| date | Fri, 05 Nov 2010 20:59:13 -0700 |
| parents | c808684660a7 |
| children | 54aa96ae8912 |
| 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 |
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17 #ifdef ANDROID_NDK |
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18 #include <AL/al.h> |
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19 #include <AL/alc.h> |
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20 #else |
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21 #include "al.h" /* OpenAL */ |
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22 #include "alc.h" /* For creating OpenAL contexts */ |
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23 #endif |
| 0 | 24 |
| 1 | 25 #ifdef __APPLE__ |
| 26 /* For performance things like ALC_CONVERT_DATA_UPON_LOADING */ | |
| 27 /* Note: ALC_CONVERT_DATA_UPON_LOADING used to be in the alc.h header. | |
| 28 * But in the Tiger OpenAL 1.1 release (10.4.7 and Xcode 2.4), the | |
| 29 * define was moved to a new header file and renamed to | |
| 30 * ALC_MAC_OSX_CONVERT_DATA_UPON_LOADING. | |
| 31 */ | |
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32 /* |
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33 #include <TargetConditionals.h> |
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34 #if (TARGET_OS_IPHONE == 1) || (TARGET_IPHONE_SIMULATOR == 1) |
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35 |
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36 #else |
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37 #include <OpenAL/MacOSX_OALExtensions.h> |
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38 #endif |
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39 */ |
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40 |
| 1 | 41 #endif |
| 42 | |
| 0 | 43 /* For malloc, bsearch, qsort */ |
| 44 #include <stdlib.h> | |
| 45 | |
| 46 /* For memcpy */ | |
| 47 #include <string.h> | |
| 48 | |
| 49 #if 0 | |
| 50 /* for toupper */ | |
| 51 #include <ctype.h> | |
| 52 /* for strrchr */ | |
| 53 #include <string.h> | |
| 54 #endif | |
| 55 | |
| 56 /* Currently used in the output debug functions */ | |
| 57 #include <stdio.h> | |
| 58 | |
| 59 /* My own CircularQueue implementation needed | |
| 60 * to work around the Nvidia problem of the | |
| 61 * lack of a buffer query. | |
| 62 */ | |
| 63 #include "CircularQueue.h" | |
| 64 | |
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65 #ifdef ENABLE_ALMIXER_THREADS |
| 0 | 66 /* Needed for the Mutex locks (and threads if enabled) */ |
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67 #ifdef ALMIXER_COMPILE_WITHOUT_SDL |
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68 #include "SimpleMutex.h" |
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69 #include "SimpleThread.h" |
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70 typedef struct SimpleMutex SDL_mutex; |
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71 typedef struct SimpleThread SDL_Thread; |
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72 #define SDL_CreateMutex SimpleMutex_CreateMutex |
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73 #define SDL_DestroyMutex SimpleMutex_DestroyMutex |
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74 #define SDL_LockMutex SimpleMutex_LockMutex |
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75 #define SDL_UnlockMutex SimpleMutex_UnlockMutex |
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76 #define SDL_CreateThread SimpleThread_CreateThread |
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77 #define SDL_WaitThread SimpleThread_WaitThread |
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78 |
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79 #else |
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80 #include "SDL_thread.h" |
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81 #endif |
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82 #endif |
| 0 | 83 |
| 84 /* Because of the API differences between the Loki | |
| 85 * and Creative distributions, we need to know which | |
| 86 * version to use. The LOKI distribution currently | |
| 87 * has AL_BYTE_LOKI defined in altypes.h which | |
| 88 * I will use as a flag to identify the distributions. | |
| 89 * If this is ever removed, I might revert back to the | |
| 90 * if defined(_WIN32) or defined(__APPLE__) test to | |
| 91 * identify the Creative dist. | |
| 92 * I'm not sure if or how the Nvidia distribution differs | |
| 93 * from the Creative distribution. So for | |
| 94 * now, the Nvidia distribution gets lumped with the | |
| 95 * Creative dist and I hope nothing will break. | |
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96 * My alGetString may be the most vulnerable. |
| 0 | 97 */ |
| 98 #ifdef AL_BYTE_LOKI | |
| 99 #define USING_LOKI_AL_DIST | |
| 100 /* This is a short term fix to get around the | |
| 101 * queuing problem with non-power of two buffer sizes. | |
| 102 * Hopefully the maintainers will fix this before | |
| 103 * we're ready to ship. | |
| 104 */ | |
| 105 #define ENABLE_LOKI_QUEUE_FIX_HACK | |
| 106 | |
| 107 /* The AL_GAIN in the Loki dist doesn't seem to do | |
| 108 * what I want/expect it to do. I want to use it for | |
| 109 * Fading, but it seems to work like an off/on switch. | |
| 110 * 0 = off, >0 = on. | |
| 111 * The AL_GAIN_LINEAR_LOKI switch seems to do what | |
| 112 * I want, so I'll redefine it here so the code is consistent | |
| 113 */ | |
| 114 /* Update: I've changed the source volume implementations | |
| 115 * to use AL_MAX_GAIN, so I don't think I need this block | |
| 116 * of code anymore. The listener uses AL_GAIN, but I | |
| 117 * hope they got this one right since there isn't a AL_MAX_GAIN | |
| 118 * for the listener. | |
| 119 */ | |
| 120 /* | |
| 121 #undef AL_GAIN | |
| 122 #include "alexttypes.h" | |
| 123 #define AL_GAIN AL_GAIN_LINEAR_LOKI | |
| 124 */ | |
| 125 #else | |
| 126 /* Might need to run other tests to figure out the DIST */ | |
| 127 /* I've been told that Nvidia doesn't define constants | |
| 128 * in the headers like Creative. Instead of | |
| 129 * #define AL_REFERENCE_DISTANCE 0x1020, | |
| 130 * Nvidia prefers you query OpenAL for a value. | |
| 131 * int AL_REFERENCE_DISTANCE = alGetEnumValue(ALubyte*)"AL_REFERNECE_DISTANCE"); | |
| 132 * So I'm assuming this means the Nvidia lacks this value. | |
| 133 * If this is the case, | |
| 134 * I guess we can use it to identify the Nvidia dist | |
| 135 */ | |
| 136 #ifdef AL_REFERENCE_DISTANCE | |
| 137 #define USING_CREATIVE_AL_DIST | |
| 138 #else | |
| 139 #define USING_NVIDIA_AL_DIST | |
| 140 #endif | |
| 141 #endif | |
| 142 | |
| 143 #ifdef ENABLE_LOKI_QUEUE_FIX_HACK | |
| 144 /* Need memset to zero out data */ | |
| 145 #include <string.h> | |
| 146 #endif | |
| 147 | |
| 148 | |
| 149 /* Seek issues for predecoded samples: | |
| 150 * The problem is that OpenAL makes us copy an | |
| 151 * entire buffer if we want to use it. This | |
| 152 * means we potentially have two copies of the | |
| 153 * same data. For predecoded data, this can be a | |
| 154 * large amount of memory. However, for seek | |
| 155 * support, I need to be able to get access to | |
| 156 * the original data so I can set byte positions. | |
| 157 * The following flags let you disable seek support | |
| 158 * if you don't want the memory hit, keep everything, | |
| 159 * or let you try to minimize the memory wasted by | |
| 160 * fetching it from the OpenAL buffer if needed | |
| 161 * and making a copy of it. | |
| 162 * Update: I don't think I need this flag anymore. I've made the | |
| 163 * effects of this user customizable by the access_data flag on load. | |
| 164 * If set to true, then seek and data callbacks work, with the | |
| 165 * cost of more memory and possibly CPU for copying the data through | |
| 166 * the callbacks. If false, then the extra memory is freed, but | |
| 167 * you don't get the features. | |
| 168 */ | |
| 169 /* | |
| 170 #define DISABLE_PREDECODED_SEEK | |
| 171 */ | |
| 172 /* Problem: Even though alGetBufferi(., AL_DATA, .) | |
| 173 * is in the Creative Programmer's reference, | |
| 174 * it actually isn't in the dist. (Invalid enum | |
| 175 * in Creative, can't compile in Loki.) | |
| 176 * So we have to keep it disabled | |
| 177 */ | |
| 178 #define DISABLE_SEEK_MEMORY_OPTIMIZATION | |
| 179 | |
| 180 #ifndef DISABLE_SEEK_MEMORY_OPTIMIZATION | |
| 181 /* Needed for memcpy */ | |
| 182 #include <string.h> | |
| 183 #endif | |
| 184 | |
| 185 /* Old way of doing things: | |
| 186 #if defined(_WIN32) || defined(__APPLE__) | |
| 187 #define USING_CREATIVE_AL_DIST | |
| 188 #else | |
| 189 #define USING_LOKI_AL_DIST | |
| 190 #endif | |
| 191 */ | |
| 192 | |
| 193 /************ REMOVE ME (Don't need anymore) ********/ | |
| 194 #if 0 | |
| 195 /* Let's get fancy and see if triple buffering | |
| 196 * does anything good for us | |
| 197 * Must be 2 or more or things will probably break | |
| 198 */ | |
| 199 #define NUMBER_OF_QUEUE_BUFFERS 5 | |
| 200 /* This is the number of buffers that are queued up | |
| 201 * when play first starts up. This should be at least 1 | |
| 202 * and no more than NUMBER_OF_QUEUE_BUFFERS | |
| 203 */ | |
| 204 #define NUMBER_OF_START_UP_BUFFERS 2 | |
| 205 #endif | |
| 206 /************ END REMOVE ME (Don't need anymore) ********/ | |
| 207 | |
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208 #ifdef ALMIXER_COMPILE_WITHOUT_SDL |
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209 #include "tErrorLib.h" |
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210 static TErrorPool* s_ALmixerErrorPool = NULL; |
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211 #endif |
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212 |
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213 static ALboolean ALmixer_Initialized = 0; |
| 0 | 214 /* This should be set correctly by Init */ |
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215 static ALuint ALmixer_Frequency_global = ALMIXER_DEFAULT_FREQUENCY; |
| 0 | 216 |
| 217 /* Will be initialized in Init */ | |
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218 static ALint Number_of_Channels_global = 0; |
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219 static ALint Number_of_Reserve_Channels_global = 0; |
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220 static ALuint Is_Playing_global = 0; |
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221 |
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222 #ifdef ENABLE_ALMIXER_THREADS |
| 0 | 223 /* This is for a simple lock system. It is not meant to be good, |
| 224 * but just sufficient to minimize/avoid threading issues | |
| 225 */ | |
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226 static SDL_mutex* s_simpleLock; |
| 0 | 227 static SDL_Thread* Stream_Thread_global = NULL; |
| 228 #endif | |
| 229 | |
| 230 | |
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231 #ifdef __APPLE__ |
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232 static ALvoid Internal_alcMacOSXMixerOutputRate(const ALdouble sample_rate) |
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233 { |
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234 static void (*alcMacOSXMixerOutputRateProcPtr)(const ALdouble) = NULL; |
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235 |
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236 if(NULL == alcMacOSXMixerOutputRateProcPtr) |
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237 { |
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238 alcMacOSXMixerOutputRateProcPtr = alGetProcAddress((const ALCchar*) "alcMacOSXMixerOutputRate"); |
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239 } |
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240 |
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241 if(NULL != alcMacOSXMixerOutputRateProcPtr) |
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242 { |
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243 alcMacOSXMixerOutputRateProcPtr(sample_rate); |
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244 } |
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245 |
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246 return; |
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247 } |
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248 |
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249 ALdouble Internal_alcMacOSXGetMixerOutputRate() |
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250 { |
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251 static ALdouble (*alcMacOSXGetMixerOutputRateProcPtr)(void) = NULL; |
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252 |
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253 if(NULL == alcMacOSXGetMixerOutputRateProcPtr) |
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254 { |
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255 alcMacOSXGetMixerOutputRateProcPtr = alGetProcAddress((const ALCchar*) "alcMacOSXGetMixerOutputRate"); |
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256 } |
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257 |
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258 if(NULL != alcMacOSXGetMixerOutputRateProcPtr) |
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259 { |
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260 return alcMacOSXGetMixerOutputRateProcPtr(); |
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261 } |
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262 |
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263 return 0.0; |
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264 } |
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265 #endif |
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266 |
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267 #ifdef ALMIXER_COMPILE_WITHOUT_SDL |
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268 |
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269 #if defined(__APPLE__) |
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270 #include <QuartzCore/QuartzCore.h> |
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271 #include <unistd.h> |
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272 static CFTimeInterval s_ticksBaseTime = 0.0; |
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273 |
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274 #elif defined(_WIN32) |
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275 #define WIN32_LEAN_AND_MEAN |
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276 #include <windows.h> |
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277 #include <winbase.h> |
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278 LARGE_INTEGER s_hiResTicksPerSecond; |
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279 double s_hiResSecondsPerTick; |
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280 LARGE_INTEGER s_ticksBaseTime; |
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281 #else |
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282 #include <unistd.h> |
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283 #include <time.h> |
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284 static struct timespec s_ticksBaseTime; |
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285 #endif |
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286 static void ALmixer_InitTime() |
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287 { |
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288 #if defined(__APPLE__) |
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289 s_ticksBaseTime = CACurrentMediaTime(); |
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290 |
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291 #elif defined(_WIN32) |
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292 LARGE_INTEGER hi_res_ticks_per_second; |
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293 if(TRUE == QueryPerformanceFrequency(&hi_res_ticks_per_second)) |
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294 { |
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295 QueryPerformanceCounter(&s_ticksBaseTime); |
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296 s_hiResSecondsPerTick = 1.0 / hi_res_ticks_per_second; |
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297 } |
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298 else |
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299 { |
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300 ALMixer_SetError("Windows error: High resolution clock failed."); |
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301 fprintf(stderr, "Windows error: High resolution clock failed. Audio will not work correctly.\n"); |
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302 } |
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303 #else |
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304 /* clock_gettime is POSIX.1-2001 */ |
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305 clock_gettime(CLOCK_MONOTONIC, &s_ticksBaseTime); |
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306 #endif |
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307 |
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308 } |
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309 static ALuint ALmixer_GetTicks() |
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310 { |
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311 #if defined(__APPLE__) |
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312 return (ALuint)((CACurrentMediaTime()-s_ticksBaseTime)*1000.0); |
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313 #elif defined(_WIN32) |
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314 LARGE_INTEGER current_time; |
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315 QueryPerformanceCounter(¤t_time); |
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316 return (ALuint)((current_time.QuadPart - s_ticksBaseTime.QuadPart) * 1000 * s_hiResSecondsPerTick); |
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317 #elif ANDROID_NDK |
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318 |
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319 #else /* assuming POSIX */ |
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320 /* clock_gettime is POSIX.1-2001 */ |
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321 struct timespec current_time; |
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322 clock_gettime(CLOCK_MONOTONIC, ¤t_time); |
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323 return (ALuint)((current_time.tv_sec - s_ticksBaseTime.tv_sec)*1000.0 + (current_time.tv_nsec - s_ticksBaseTime.tv_nsec) / 1000000); |
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324 #endif |
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325 } |
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326 static void ALmixer_Delay(ALuint milliseconds_delay) |
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327 { |
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328 #if defined(_WIN32) |
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329 Sleep(milliseconds_delay); |
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330 #else |
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331 usleep(milliseconds_delay); |
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332 #endif |
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333 } |
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334 #else |
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335 #include "SDL.h" /* For SDL_GetTicks(), SDL_Delay */ |
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336 #define ALmixer_GetTicks SDL_GetTicks |
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337 #define ALmixer_Delay SDL_Delay |
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338 #endif |
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339 |
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340 |
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341 |
| 0 | 342 /* If ENABLE_PARANOID_SIGNEDNESS_CHECK is used, |
| 343 * these values will be reset on Init() | |
| 344 * Consider these values Read-Only. | |
| 345 */ | |
| 346 | |
| 347 #define ALMIXER_SIGNED_VALUE 127 | |
| 348 #define ALMIXER_UNSIGNED_VALUE 255 | |
| 349 | |
| 350 #ifdef ENABLE_PARANOID_SIGNEDNESS_CHECK | |
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351 static ALushort SIGN_TYPE_16_BIT_FORMAT = AUDIO_S16SYS; |
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352 static ALushort SIGN_TYPE_8_BIT_FORMAT = AUDIO_S8; |
| 0 | 353 #else |
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354 static const ALushort SIGN_TYPE_16_BIT_FORMAT = AUDIO_S16SYS; |
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355 static const ALushort SIGN_TYPE_8_BIT_FORMAT = AUDIO_S8; |
| 0 | 356 #endif |
| 357 | |
| 1 | 358 |
| 359 /* This can be private instead of being in the header now that I moved | |
| 360 * ALmixer_Data inside here. | |
| 361 */ | |
| 362 typedef struct ALmixer_Buffer_Map ALmixer_Buffer_Map; | |
| 363 | |
| 364 | |
| 365 struct ALmixer_Data | |
| 366 { | |
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367 ALboolean decoded_all; /* dictates different behaviors */ |
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368 ALint total_time; /* total playing time of sample (msec) */ |
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369 |
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370 ALuint in_use; /* needed to prevent sharing for streams */ |
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371 ALboolean eof; /* flag for eof, only used for streams */ |
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372 |
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373 ALuint total_bytes; /* For predecoded */ |
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374 ALuint loaded_bytes; /* For predecoded (for seek) */ |
| 1 | 375 |
| 376 Sound_Sample* sample; /* SDL_Sound provides the data */ | |
| 377 ALuint* buffer; /* array of OpenAL buffers (at least 1 for predecoded) */ | |
| 378 | |
| 379 /* Needed for streamed buffers */ | |
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380 ALuint max_queue_buffers; /* Max number of queue buffers */ |
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381 ALuint num_startup_buffers; /* Number of ramp-up buffers */ |
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382 ALuint num_buffers_in_use; /* number of buffers in use */ |
| 1 | 383 |
| 384 /* This stuff is for streamed buffers that require data access */ | |
| 385 ALmixer_Buffer_Map* buffer_map_list; /* translate ALbuffer to index | |
| 386 and holds pointer to copy of data for | |
| 387 data access */ | |
| 388 ALuint current_buffer; /* The current playing buffer */ | |
| 389 | |
| 390 /* Nvidia distribution refuses to recognize a simple buffer query command | |
| 391 * unlike all other distributions. It's forcing me to redo the code | |
| 392 * to accomodate this Nvidia flaw by making me maintain a "best guess" | |
| 393 * copy of what I think the buffer queue state looks like. | |
| 394 * A circular queue would a helpful data structure for this task, | |
| 395 * but I wanted to avoid making an additional header requirement, | |
| 396 * so I'm making it a void* | |
| 397 */ | |
| 398 void* circular_buffer_queue; | |
| 399 | |
| 400 | |
| 401 }; | |
| 402 | |
| 0 | 403 static struct ALmixer_Channel |
| 404 { | |
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405 ALboolean channel_in_use; |
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406 ALboolean callback_update; /* For streaming determination */ |
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407 ALboolean needs_stream; /* For streaming determination */ |
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408 ALboolean halted; |
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409 ALboolean paused; |
| 0 | 410 ALuint alsource; |
| 411 ALmixer_Data* almixer_data; | |
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412 ALint loops; |
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413 ALint expire_ticks; |
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414 ALuint start_time; |
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415 |
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416 ALboolean fade_enabled; |
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417 ALuint fade_expire_ticks; |
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418 ALuint fade_start_time; |
| 0 | 419 ALfloat fade_inv_time; |
| 420 ALfloat fade_start_volume; | |
| 421 ALfloat fade_end_volume; | |
| 422 ALfloat max_volume; | |
| 423 ALfloat min_volume; | |
| 424 | |
| 425 /* Do we need other flags? | |
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426 ALbyte *samples; |
| 0 | 427 int volume; |
| 428 int looping; | |
| 429 int tag; | |
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430 ALuint expire; |
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431 ALuint start_time; |
| 0 | 432 Mix_Fading fading; |
| 433 int fade_volume; | |
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434 ALuint fade_length; |
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435 ALuint ticks_fade; |
| 0 | 436 effect_info *effects; |
| 437 */ | |
| 438 } *ALmixer_Channel_List = NULL; | |
| 439 | |
| 1 | 440 struct ALmixer_Buffer_Map |
| 441 { | |
| 442 ALuint albuffer; | |
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443 ALint index; /* might not need */ |
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444 ALbyte* data; |
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445 ALuint num_bytes; |
| 1 | 446 }; |
| 447 | |
| 0 | 448 /* This will be used to find a channel if the user supplies a source */ |
| 449 typedef struct Source_Map | |
| 450 { | |
| 451 ALuint source; | |
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452 ALint channel; |
| 0 | 453 } Source_Map; |
| 454 /* Keep an array of all sources with their associated channel */ | |
| 455 static Source_Map* Source_Map_List; | |
| 456 | |
| 457 static int Compare_Source_Map(const void* a, const void* b) | |
| 458 { | |
| 459 return ( ((Source_Map*)a)->source - ((Source_Map*)b)->source ); | |
| 460 } | |
| 461 | |
| 462 /* Sort by channel instead of source */ | |
| 463 static int Compare_Source_Map_by_channel(const void* a, const void* b) | |
| 464 { | |
| 465 return ( ((Source_Map*)a)->channel - ((Source_Map*)b)->channel ); | |
| 466 } | |
| 467 | |
| 468 /* Compare by albuffer */ | |
| 469 static int Compare_Buffer_Map(const void* a, const void* b) | |
| 470 { | |
| 1 | 471 return ( ((ALmixer_Buffer_Map*)a)->albuffer - ((ALmixer_Buffer_Map*)b)->albuffer ); |
| 0 | 472 } |
| 473 | |
| 474 /* This is for the user defined callback via | |
| 475 * ALmixer_ChannelFinished() | |
| 476 */ | |
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477 static void (*Channel_Done_Callback)(ALint which_channel, ALuint al_source, ALmixer_Data* almixer_data, ALboolean finished_naturally, void* user_data) = NULL; |
| 0 | 478 static void* Channel_Done_Callback_Userdata = NULL; |
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479 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 | 480 static void* Channel_Data_Callback_Userdata = NULL; |
| 0 | 481 |
| 482 | |
| 483 static void PrintQueueStatus(ALuint source) | |
| 484 { | |
| 485 ALint buffers_queued = 0; | |
| 486 ALint buffers_processed = 0; | |
| 487 ALenum error; | |
| 488 | |
| 489 /* Get the number of buffers still queued */ | |
| 490 alGetSourcei( | |
| 491 source, | |
| 492 AL_BUFFERS_QUEUED, | |
| 493 &buffers_queued | |
| 494 ); | |
| 495 | |
| 496 if((error = alGetError()) != AL_NO_ERROR) | |
| 497 { | |
| 498 fprintf(stderr, "Error in PrintQueueStatus, Can't get buffers_queued: %s\n", | |
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499 alGetString(error)); |
| 0 | 500 } |
| 501 /* Get the number of buffers processed | |
| 502 * so we know if we need to refill | |
| 503 */ | |
| 504 alGetSourcei( | |
| 505 source, | |
| 506 AL_BUFFERS_PROCESSED, | |
| 507 &buffers_processed | |
| 508 ); | |
| 509 if((error = alGetError()) != AL_NO_ERROR) | |
| 510 { | |
| 511 fprintf(stderr, "Error in PrintQueueStatus, Can't get buffers_processed: %s\n", | |
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512 alGetString(error)); |
| 0 | 513 } |
| 514 | |
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515 /* |
| 0 | 516 fprintf(stderr, "For source: %d, buffers_queued=%d, buffers_processed=%d\n", |
| 517 source, | |
| 518 buffers_queued, | |
| 519 buffers_processed); | |
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520 */ |
| 0 | 521 } |
| 522 | |
| 523 | |
| 524 | |
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525 static void Init_Channel(ALint channel) |
| 0 | 526 { |
| 527 ALmixer_Channel_List[channel].channel_in_use = 0; | |
| 528 ALmixer_Channel_List[channel].callback_update = 0; | |
| 529 ALmixer_Channel_List[channel].needs_stream = 0; | |
| 530 ALmixer_Channel_List[channel].paused = 0; | |
| 531 ALmixer_Channel_List[channel].halted = 0; | |
| 532 ALmixer_Channel_List[channel].loops = 0; | |
| 533 | |
| 534 ALmixer_Channel_List[channel].expire_ticks = 0; | |
| 535 ALmixer_Channel_List[channel].start_time = 0; | |
| 536 | |
| 537 ALmixer_Channel_List[channel].fade_enabled = 0; | |
| 538 ALmixer_Channel_List[channel].fade_expire_ticks = 0; | |
| 539 ALmixer_Channel_List[channel].fade_start_time = 0; | |
| 540 ALmixer_Channel_List[channel].fade_inv_time = 0.0f; | |
| 541 ALmixer_Channel_List[channel].fade_start_volume = 0.0f; | |
| 542 ALmixer_Channel_List[channel].fade_end_volume = 0.0f; | |
| 543 ALmixer_Channel_List[channel].max_volume = 1.0f; | |
| 544 ALmixer_Channel_List[channel].min_volume = 0.0f; | |
| 545 | |
| 546 ALmixer_Channel_List[channel].almixer_data = NULL; | |
| 547 } | |
| 548 /* Quick helper function to clean up a channel | |
| 549 * after it's done playing */ | |
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550 static void Clean_Channel(ALint channel) |
| 0 | 551 { |
| 552 ALenum error; | |
| 553 ALmixer_Channel_List[channel].channel_in_use = 0; | |
| 554 ALmixer_Channel_List[channel].callback_update = 0; | |
| 555 ALmixer_Channel_List[channel].needs_stream = 0; | |
| 556 ALmixer_Channel_List[channel].paused = 0; | |
| 557 ALmixer_Channel_List[channel].halted = 0; | |
| 558 ALmixer_Channel_List[channel].loops = 0; | |
| 559 | |
| 560 | |
| 561 ALmixer_Channel_List[channel].expire_ticks = 0; | |
| 562 ALmixer_Channel_List[channel].start_time = 0; | |
| 563 | |
| 564 ALmixer_Channel_List[channel].fade_enabled = 0; | |
| 565 ALmixer_Channel_List[channel].fade_expire_ticks = 0; | |
| 566 ALmixer_Channel_List[channel].fade_start_time = 0; | |
| 567 ALmixer_Channel_List[channel].fade_inv_time = 0.0f; | |
| 568 ALmixer_Channel_List[channel].fade_start_volume = 0.0f; | |
| 569 ALmixer_Channel_List[channel].fade_end_volume = 0.0f; | |
| 570 | |
| 571 alSourcef(ALmixer_Channel_List[channel].alsource, AL_MAX_GAIN, | |
| 572 ALmixer_Channel_List[channel].max_volume); | |
| 573 | |
| 574 if((error = alGetError()) != AL_NO_ERROR) | |
| 575 { | |
| 576 fprintf(stderr, "10Testing errpr before unqueue because getting stuff, for OS X this is expected: %s\n", | |
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577 alGetString(error)); |
| 0 | 578 } |
| 579 | |
| 580 alSourcef(ALmixer_Channel_List[channel].alsource, AL_MIN_GAIN, | |
| 581 ALmixer_Channel_List[channel].min_volume); | |
| 582 if((error = alGetError()) != AL_NO_ERROR) | |
| 583 { | |
| 584 fprintf(stderr, "11Testing errpr before unqueue because getting stuff, for OS X this is expected: %s\n", | |
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585 alGetString(error)); |
| 0 | 586 } |
| 587 | |
| 588 if(ALmixer_Channel_List[channel].almixer_data != NULL) | |
| 589 { | |
| 590 if(ALmixer_Channel_List[channel].almixer_data->in_use > 0) | |
| 591 { | |
| 592 ALmixer_Channel_List[channel].almixer_data->in_use--; | |
| 593 } | |
| 594 } | |
| 595 /* Needed to determine if rewind is needed, can't reset */ | |
| 596 /* | |
| 597 ALmixer_Channel_List[channel].almixer_data->eof = 0; | |
| 598 */ | |
| 599 | |
| 600 ALmixer_Channel_List[channel].almixer_data = NULL; | |
| 601 } | |
| 602 | |
| 603 | |
| 604 #if 0 | |
| 605 /* Not needed anymore because not doing any fileext checks. | |
| 606 * | |
| 607 * Unfortunately, strcasecmp isn't portable so here's a | |
| 608 * reimplementation of it (taken from SDL_sound) | |
| 609 */ | |
| 610 static int ALmixer_strcasecmp(const char* x, const char* y) | |
| 611 { | |
| 612 int ux, uy; | |
| 613 | |
| 614 if (x == y) /* same pointer? Both NULL? */ | |
| 615 return(0); | |
| 616 | |
| 617 if (x == NULL) | |
| 618 return(-1); | |
| 619 | |
| 620 if (y == NULL) | |
| 621 return(1); | |
| 622 | |
| 623 do | |
| 624 { | |
| 625 ux = toupper((int) *x); | |
| 626 uy = toupper((int) *y); | |
| 627 if (ux > uy) | |
| 628 return(1); | |
| 629 else if (ux < uy) | |
| 630 return(-1); | |
| 631 x++; | |
| 632 y++; | |
| 633 } while ((ux) && (uy)); | |
| 634 | |
| 635 return(0); | |
| 636 } | |
| 637 #endif | |
| 638 | |
| 639 | |
| 640 /* What shoud this return? | |
| 641 * 127 for signed, 255 for unsigned | |
| 642 */ | |
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643 static ALubyte GetSignednessValue(ALushort format) |
| 0 | 644 { |
| 645 switch(format) | |
| 646 { | |
| 647 case AUDIO_U8: | |
| 648 case AUDIO_U16LSB: | |
| 649 case AUDIO_U16MSB: | |
| 650 return ALMIXER_UNSIGNED_VALUE; | |
| 651 break; | |
| 652 case AUDIO_S8: | |
| 653 case AUDIO_S16LSB: | |
| 654 case AUDIO_S16MSB: | |
| 655 return ALMIXER_SIGNED_VALUE; | |
| 656 break; | |
| 657 default: | |
| 658 return 0; | |
| 659 } | |
| 660 return 0; | |
| 661 } | |
| 662 | |
| 663 | |
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664 static ALubyte GetBitDepth(ALushort format) |
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665 { |
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666 ALubyte bit_depth = 16; |
| 0 | 667 |
| 668 switch(format) | |
| 669 { | |
| 670 case AUDIO_U8: | |
| 671 case AUDIO_S8: | |
| 672 bit_depth = 8; | |
| 673 break; | |
| 674 | |
| 675 case AUDIO_U16LSB: | |
| 676 /* | |
| 677 case AUDIO_U16: | |
| 678 */ | |
| 679 case AUDIO_S16LSB: | |
| 680 /* | |
| 681 case AUDIO_S16: | |
| 682 */ | |
| 683 case AUDIO_U16MSB: | |
| 684 case AUDIO_S16MSB: | |
| 685 /* | |
| 686 case AUDIO_U16SYS: | |
| 687 case AUDIO_S16SYS: | |
| 688 */ | |
| 689 bit_depth = 16; | |
| 690 break; | |
| 691 default: | |
| 692 bit_depth = 0; | |
| 693 } | |
| 694 return bit_depth; | |
| 695 } | |
| 696 | |
| 697 /* Need to translate between SDL/SDL_Sound audiospec | |
| 698 * and OpenAL conventions */ | |
| 699 static ALenum TranslateFormat(Sound_AudioInfo* info) | |
| 700 { | |
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701 ALubyte bit_depth; |
| 0 | 702 |
| 703 bit_depth = GetBitDepth(info->format); | |
| 704 if(0 == bit_depth) | |
| 705 { | |
| 706 fprintf(stderr, "Warning: Unknown bit depth. Setting to 16\n"); | |
| 707 bit_depth = 16; | |
| 708 } | |
| 709 | |
| 710 if(2 == info->channels) | |
| 711 { | |
| 712 if(16 == bit_depth) | |
| 713 { | |
| 714 return AL_FORMAT_STEREO16; | |
| 715 } | |
| 716 else | |
| 717 { | |
| 718 return AL_FORMAT_STEREO8; | |
| 719 } | |
| 720 } | |
| 721 else | |
| 722 { | |
| 723 if(16 == bit_depth) | |
| 724 { | |
| 725 return AL_FORMAT_MONO16; | |
| 726 } | |
| 727 else | |
| 728 { | |
| 729 return AL_FORMAT_MONO8; | |
| 730 } | |
| 731 } | |
| 732 /* Make compiler happy. Shouldn't get here */ | |
| 733 return AL_FORMAT_STEREO16; | |
| 734 } | |
| 735 | |
| 1 | 736 |
| 737 /* This will compute the total playing time | |
| 738 * based upon the number of bytes and audio info. | |
| 739 * (In prinicple, it should compute the time for any given length) | |
| 740 */ | |
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741 static ALuint Compute_Total_Time_Decomposed(ALuint bytes_per_sample, ALuint frequency, ALubyte channels, size_t total_bytes) |
| 1 | 742 { |
| 743 double total_sec; | |
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744 ALuint total_msec; |
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745 ALuint bytes_per_sec; |
| 1 | 746 |
| 747 if(0 == total_bytes) | |
| 748 { | |
| 749 return 0; | |
| 750 } | |
| 751 /* To compute Bytes per second, do | |
| 752 * samples_per_sec * bytes_per_sample * number_of_channels | |
| 753 */ | |
| 754 bytes_per_sec = frequency * bytes_per_sample * channels; | |
| 755 | |
| 756 /* Now to get total time (sec), do | |
| 757 * total_bytes / bytes_per_sec | |
| 758 */ | |
| 759 total_sec = total_bytes / (double)bytes_per_sec; | |
| 760 | |
| 761 /* Now convert seconds to milliseconds | |
| 762 * Add .5 to the float to do rounding before the final cast | |
| 763 */ | |
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764 total_msec = (ALuint) ( (total_sec * 1000) + 0.5 ); |
| 1 | 765 /* |
| 766 fprintf(stderr, "freq=%d, bytes_per_sample=%d, channels=%d, total_msec=%d\n", frequency, bytes_per_sample, channels, total_msec); | |
| 767 */ | |
| 768 return total_msec; | |
| 769 } | |
| 770 | |
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771 static ALuint Compute_Total_Time(Sound_AudioInfo *info, size_t total_bytes) |
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772 { |
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773 ALuint bytes_per_sample; |
| 1 | 774 |
| 775 if(0 == total_bytes) | |
| 776 { | |
| 777 return 0; | |
| 778 } | |
| 779 /* SDL has a mask trick I was not aware of. Mask the upper bits | |
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780 * of the format, and you get 8 or 16 which is the bits per sample. |
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781 * Divide by 8bits_per_bytes and you get bytes_per_sample |
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782 * I tested this under 32-bit and 64-bit and big and little endian |
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783 * to make sure this still works since I have since moved from |
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784 * Uint32 to unspecified size types like ALuint. |
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785 */ |
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786 bytes_per_sample = (ALuint) ((info->format & 0xFF) / 8); |
| 1 | 787 |
| 788 return Compute_Total_Time_Decomposed(bytes_per_sample, info->rate, info->channels, total_bytes); | |
| 789 } /* End Compute_Total_Time */ | |
| 790 | |
| 791 | |
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792 static size_t Compute_Total_Bytes_Decomposed(ALuint bytes_per_sample, ALuint frequency, ALubyte channels, ALuint total_msec) |
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793 { |
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794 double total_sec; |
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795 ALuint bytes_per_sec; |
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796 size_t total_bytes; |
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797 |
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798 if(0 >= total_msec) |
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799 { |
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800 return 0; |
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801 } |
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802 /* To compute Bytes per second, do |
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803 * samples_per_sec * bytes_per_sample * number_of_channels |
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804 */ |
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805 bytes_per_sec = frequency * bytes_per_sample * channels; |
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806 |
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807 /* convert milliseconds to seconds */ |
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808 total_sec = total_msec / 1000.0; |
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809 |
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810 /* Now to get total bytes */ |
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811 total_bytes = (size_t)(((double)bytes_per_sec * total_sec) + 0.5); |
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812 |
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813 /* 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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814 */ |
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815 |
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816 return total_bytes; |
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817 } |
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818 |
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819 static size_t Compute_Total_Bytes(Sound_AudioInfo *info, ALuint total_msec) |
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820 { |
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821 ALuint bytes_per_sample; |
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822 |
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823 if(0 >= total_msec) |
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824 { |
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825 return 0; |
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826 } |
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827 /* SDL has a mask trick I was not aware of. Mask the upper bits |
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828 * of the format, and you get 8 or 16 which is the bits per sample. |
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829 * Divide by 8bits_per_bytes and you get bytes_per_sample |
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830 * I tested this under 32-bit and 64-bit and big and little endian |
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831 * to make sure this still works since I have since moved from |
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832 * Uint32 to unspecified size types like ALuint. |
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833 */ |
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834 bytes_per_sample = (ALuint) ((info->format & 0xFF) / 8); |
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835 |
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836 return Compute_Total_Bytes_Decomposed(bytes_per_sample, info->rate, info->channels, total_msec); |
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837 } |
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838 |
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839 /* The back-end decoders seem to need to decode in quantized frame sizes. |
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840 * So if I can pad the bytes to the next quanta, things might go more smoothly. |
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841 */ |
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842 static size_t Compute_Total_Bytes_With_Frame_Padding(Sound_AudioInfo *info, ALuint total_msec) |
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843 { |
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844 ALuint bytes_per_sample; |
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845 ALuint bytes_per_frame; |
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846 size_t evenly_divisible_frames; |
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847 size_t remainder_frames; |
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848 size_t return_bytes; |
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849 |
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850 size_t total_bytes = Compute_Total_Bytes(info, total_msec); |
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851 |
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852 bytes_per_sample = (ALuint) ((info->format & 0xFF) / 8); |
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853 |
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854 bytes_per_frame = bytes_per_sample * info->channels; |
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855 |
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856 evenly_divisible_frames = total_bytes / bytes_per_frame; |
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857 remainder_frames = total_bytes % bytes_per_frame; |
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858 |
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859 return_bytes = (evenly_divisible_frames * bytes_per_frame) + (remainder_frames * bytes_per_frame); |
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860 |
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861 /* Experimentally, some times I see to come up short in |
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862 * actual bytes decoded and I see a second pass is needed. |
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863 * I'm worried this may have additional performance implications. |
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864 * Sometimes in the second pass (depending on file), |
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865 * I have seen between 0 and 18 bytes. |
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866 * I'm tempted to pad the bytes by some arbitrary amount. |
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867 * However, I think currently the way SDL_sound is implemented, |
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868 * there is a big waste of memory up front instead of per-pass, |
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869 * so maybe I shouldn't worry about this. |
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870 */ |
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871 /* |
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872 return_bytes += 64; |
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873 */ |
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874 /* |
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875 fprintf(stderr, "remainder_frames=%d, padded_total_bytes=%d\n", remainder_frames, return_bytes); |
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876 */ |
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877 return return_bytes; |
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878 |
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879 } |
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880 |
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881 |
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882 |
| 1 | 883 |
| 0 | 884 /**************** REMOVED ****************************/ |
| 885 /* This was removed because I originally thought | |
| 886 * OpenAL could return a pointer to the buffer data, | |
| 887 * but I was wrong. If something like that is ever | |
| 888 * implemented, then this might become useful. | |
| 889 */ | |
| 890 #if 0 | |
| 891 /* Reconstruct_Sound_Sample and Set_AudioInfo only | |
| 892 * are needed if the Seek memory optimization is | |
| 893 * used. Also, the Loki dist doesn't seem to support | |
| 894 * AL_DATA which I need for it. | |
| 895 */ | |
| 896 #ifndef DISABLE_SEEK_MEMORY_OPTIMIZATION | |
| 897 | |
| 898 static void Set_AudioInfo(Sound_AudioInfo* info, ALint frequency, ALint bits, ALint channels) | |
| 899 { | |
|
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900 info->rate = (ALuint)frequency; |
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901 info->channels = (ALubyte)channels; |
| 0 | 902 |
| 903 /* Not sure if it should be signed or unsigned. Hopefully | |
| 904 * that detail won't be needed. | |
| 905 */ | |
| 906 if(8 == bits) | |
| 907 { | |
| 908 info->format = AUDIO_U8; | |
| 909 } | |
| 910 else | |
| 911 { | |
| 912 info->format = AUDIO_U16SYS; | |
| 913 } | |
| 914 fprintf(stderr, "Audio info: freq=%d, chan=%d, format=%d\n", | |
| 915 info->rate, info->channels, info->format); | |
| 916 | |
| 917 } | |
| 918 | |
| 919 | |
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920 static ALint Reconstruct_Sound_Sample(ALmixer_Data* data) |
| 0 | 921 { |
| 922 ALenum error; | |
| 923 ALint* data_from_albuffer; | |
| 924 ALint freq; | |
| 925 ALint bits; | |
| 926 ALint channels; | |
| 927 ALint size; | |
| 928 | |
| 929 /* Create memory all initiallized to 0. */ | |
| 930 data->sample = (Sound_Sample*)calloc(1, sizeof(Sound_Sample)); | |
| 931 if(NULL == data->sample) | |
| 932 { | |
| 933 ALmixer_SetError("Out of memory for Sound_Sample"); | |
| 934 return -1; | |
| 935 } | |
| 936 | |
| 937 /* Clear errors */ | |
| 938 alGetError(); | |
| 939 | |
| 940 alGetBufferi(data->buffer[0], AL_FREQUENCY, &freq); | |
| 941 if((error = alGetError()) != AL_NO_ERROR) | |
| 942 { | |
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943 ALmixer_SetError("alGetBufferi(AL_FREQUENCY): %s", alGetString(error) ); |
| 0 | 944 free(data->sample); |
| 945 data->sample = NULL; | |
| 946 return -1; | |
| 947 } | |
| 948 | |
| 949 alGetBufferi(data->buffer[0], AL_BITS, &bits); | |
| 950 if((error = alGetError()) != AL_NO_ERROR) | |
| 951 { | |
|
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952 ALmixer_SetError("alGetBufferi(AL_BITS): %s", alGetString(error) ); |
| 0 | 953 free(data->sample); |
| 954 data->sample = NULL; | |
| 955 return -1; | |
| 956 } | |
| 957 | |
| 958 alGetBufferi(data->buffer[0], AL_CHANNELS, &channels); | |
| 959 if((error = alGetError()) != AL_NO_ERROR) | |
| 960 { | |
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961 ALmixer_SetError("alGetBufferi(AL_CHANNELS): %s", alGetString(error) ); |
| 0 | 962 free(data->sample); |
| 963 data->sample = NULL; | |
| 964 return -1; | |
| 965 } | |
| 966 | |
| 967 alGetBufferi(data->buffer[0], AL_SIZE, &size); | |
| 968 if((error = alGetError()) != AL_NO_ERROR) | |
| 969 { | |
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970 ALmixer_SetError("alGetBufferi(AL_SIZE): %s", alGetString(error) ); |
| 0 | 971 free(data->sample); |
| 972 data->sample = NULL; | |
| 973 return -1; | |
| 974 } | |
| 975 | |
| 976 alGetBufferi(data->buffer[0], AL_DATA, data_from_albuffer); | |
| 977 if((error = alGetError()) != AL_NO_ERROR) | |
| 978 { | |
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979 ALmixer_SetError("alGetBufferi(AL_DATA): %s", alGetString(error) ); |
| 0 | 980 free(data->sample); |
| 981 data->sample = NULL; | |
| 982 return -1; | |
| 983 } | |
| 984 | |
| 985 if(size <= 0) | |
| 986 { | |
| 987 ALmixer_SetError("No data in al buffer"); | |
| 988 free(data->sample); | |
| 989 data->sample = NULL; | |
| 990 return -1; | |
| 991 } | |
| 992 | |
| 993 /* Now that we have all the attributes, we need to | |
| 994 * allocate memory for the buffer and reconstruct | |
| 995 * the AudioInfo attributes. | |
| 996 */ | |
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997 data->sample->buffer = malloc(size*sizeof(ALbyte)); |
| 0 | 998 if(NULL == data->sample->buffer) |
| 999 { | |
| 1000 ALmixer_SetError("Out of memory for sample->buffer"); | |
| 1001 free(data->sample); | |
| 1002 data->sample = NULL; | |
| 1003 return -1; | |
| 1004 } | |
| 1005 | |
| 1006 memcpy(data->sample->buffer, data_from_albuffer, size); | |
| 1007 data->sample->buffer_size = size; | |
| 1008 | |
| 1009 /* Fill up the Sound_AudioInfo structures */ | |
| 1010 Set_AudioInfo(&data->sample->desired, freq, bits, channels); | |
| 1011 Set_AudioInfo(&data->sample->actual, freq, bits, channels); | |
| 1012 | |
| 1013 return 0; | |
| 1014 } | |
| 1015 | |
| 1016 #endif /* End DISABLE_SEEK_MEMORY_OPTIMIZATION */ | |
| 1017 #endif | |
| 1018 /*************** END REMOVED *************************/ | |
| 1019 | |
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1020 static void Invoke_Channel_Done_Callback(ALint which_channel, ALboolean did_finish_naturally) |
| 0 | 1021 { |
| 1022 if(NULL == Channel_Done_Callback) | |
| 1023 { | |
| 1024 return; | |
| 1025 } | |
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1026 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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1027 } |
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1028 |
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1029 static ALint LookUpBuffer(ALuint buffer, ALmixer_Buffer_Map* buffer_map_list, ALuint num_items_in_list) |
| 0 | 1030 { |
| 1031 /* Only the first value is used for the key */ | |
| 1 | 1032 ALmixer_Buffer_Map key = { 0, 0, NULL, 0 }; |
| 1033 ALmixer_Buffer_Map* found_item = NULL; | |
| 0 | 1034 key.albuffer = buffer; |
| 1035 | |
| 1036 /* Use the ANSI C binary search feature (yea!) */ | |
| 1 | 1037 found_item = (ALmixer_Buffer_Map*)bsearch(&key, buffer_map_list, num_items_in_list, sizeof(ALmixer_Buffer_Map), Compare_Buffer_Map); |
| 0 | 1038 if(NULL == found_item) |
| 1039 { | |
| 1040 ALmixer_SetError("Can't find buffer"); | |
| 1041 return -1; | |
| 1042 } | |
| 1043 return found_item->index; | |
| 1044 } | |
| 1045 | |
| 1046 | |
| 1047 /* FIXME: Need to pass back additional info to be useful. | |
| 1048 * Bit rate, stereo/mono (num chans), time in msec? | |
| 1049 * Precoded/streamed flag so user can plan for future data? | |
| 1050 */ | |
| 1 | 1051 /* |
| 1052 * channels: 1 for mono, 2 for stereo | |
| 1053 * | |
| 1054 */ | |
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1055 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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1056 { |
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1057 ALboolean is_unsigned; |
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1058 ALubyte bits_per_sample = GetBitDepth(format); |
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1059 ALuint bytes_per_sample; |
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1060 ALuint length_in_msec; |
| 1 | 1061 |
| 1062 if(GetSignednessValue(format) == ALMIXER_UNSIGNED_VALUE) | |
| 1063 { | |
| 1064 is_unsigned = 1; | |
| 1065 } | |
| 1066 else | |
| 1067 { | |
| 1068 is_unsigned = 0; | |
| 1069 } | |
| 1070 | |
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1071 bytes_per_sample = (ALuint) (bits_per_sample / 8); |
| 1 | 1072 |
| 1073 length_in_msec = Compute_Total_Time_Decomposed(bytes_per_sample, frequency, channels, num_bytes); | |
| 1074 | |
| 0 | 1075 /* |
| 1076 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); | |
| 1077 */ | |
| 1078 if(NULL == Channel_Data_Callback) | |
| 1079 { | |
| 1080 return; | |
| 1081 } | |
| 1 | 1082 /* |
| 1083 * Channel_Data_Callback(which_channel, data, num_bytes, frequency, channels, GetBitDepth(format), format, decode_mode_is_predecoded); | |
| 1084 */ | |
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1085 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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1086 } |
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1087 |
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1088 static void Invoke_Predecoded_Channel_Data_Callback(ALint channel, ALmixer_Data* data) |
| 0 | 1089 { |
| 1090 if(NULL == data->sample) | |
| 1091 { | |
| 1092 return; | |
| 1093 } | |
| 1094 /* The buffer position is complicated because if the current data was seeked, | |
| 1095 * we must adjust the buffer to the seek position | |
| 1096 */ | |
| 1097 Invoke_Channel_Data_Callback(channel, | |
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1098 (((ALbyte*) data->sample->buffer) + (data->total_bytes - data->loaded_bytes) ), |
| 0 | 1099 data->loaded_bytes, |
| 1100 data->sample->desired.rate, | |
| 1101 data->sample->desired.channels, | |
| 1102 data->sample->desired.format, | |
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1103 AL_TRUE |
| 0 | 1104 ); |
| 1105 } | |
| 1106 | |
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1107 static void Invoke_Streamed_Channel_Data_Callback(ALint channel, ALmixer_Data* data, ALuint buffer) |
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1108 { |
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1109 ALint index; |
| 0 | 1110 if(NULL == data->buffer_map_list) |
| 1111 { | |
| 1112 return; | |
| 1113 } | |
| 1114 index = LookUpBuffer(buffer, data->buffer_map_list, data->max_queue_buffers); | |
| 1115 /* This should catch the case where all buffers are unqueued | |
| 1116 * and the "current" buffer is id: 0 | |
| 1117 */ | |
| 1118 if(-1 == index) | |
| 1119 { | |
| 1120 return; | |
| 1121 } | |
| 1122 Invoke_Channel_Data_Callback(channel, | |
| 1123 data->buffer_map_list[index].data, | |
| 1124 data->buffer_map_list[index].num_bytes, | |
| 1125 data->sample->desired.rate, | |
| 1126 data->sample->desired.channels, | |
| 1127 data->sample->desired.format, | |
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1128 AL_FALSE |
| 0 | 1129 ); |
| 1130 } | |
| 1131 | |
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1132 /* Converts milliseconds to byte positions. |
| 0 | 1133 * This is needed for seeking on predecoded samples |
| 1134 */ | |
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1135 static ALuint Convert_Msec_To_Byte_Pos(Sound_AudioInfo *info, ALuint ms) |
| 0 | 1136 { |
| 1137 float frames_per_ms; | |
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1138 ALuint frame_offset; |
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1139 ALuint frame_size; |
| 0 | 1140 if(info == NULL) |
| 1141 { | |
| 1142 fprintf(stderr, "Error, info is NULL\n"); | |
| 1143 } | |
| 1144 | |
| 1145 /* "frames" == "sample frames" */ | |
| 1146 frames_per_ms = ((float) info->rate) / 1000.0f; | |
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1147 frame_offset = (ALuint) (frames_per_ms * ((float) ms)); |
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1148 frame_size = (ALuint) ((info->format & 0xFF) / 8) * info->channels; |
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1149 return frame_offset * frame_size; |
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1150 } |
| 0 | 1151 |
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1152 static ALint Set_Predecoded_Seek_Position(ALmixer_Data* data, ALuint byte_position) |
| 0 | 1153 { |
| 1154 ALenum error; | |
| 1155 /* clear error */ | |
| 1156 alGetError(); | |
| 1157 | |
| 1158 /* Is it greater than, or greater-than or equal to ?? */ | |
| 1159 if(byte_position > data->total_bytes) | |
| 1160 { | |
| 1161 /* We can't go past the end, so set to end? */ | |
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1162 /* |
| 0 | 1163 fprintf(stderr, "Error, can't seek past end\n"); |
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1164 */ |
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1165 |
| 0 | 1166 /* In case the below thing doesn't work, |
| 1167 * just rewind the whole thing. | |
| 1168 * | |
| 1169 alBufferData(data->buffer[0], | |
| 1170 TranslateFormat(&data->sample->desired), | |
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1171 (ALbyte*) data->sample->buffer, |
| 0 | 1172 data->total_bytes, |
| 1173 data->sample->desired.rate | |
| 1174 ); | |
| 1175 */ | |
| 1176 | |
| 1177 /* I was trying to set to the end, (1 byte remaining), | |
| 1178 * but I was getting freezes. I'm thinking it might be | |
| 1179 * another Power of 2 bug in the Loki dist. I tried 2, | |
| 1180 * and it still hung. 4 didn't hang, but I got a clip | |
| 1181 * artifact. 8 seemed to work okay. | |
| 1182 */ | |
| 1183 alBufferData(data->buffer[0], | |
| 1184 TranslateFormat(&data->sample->desired), | |
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1185 (((ALbyte*) data->sample->buffer) + (data->total_bytes - 8) ), |
| 0 | 1186 8, |
| 1187 data->sample->desired.rate | |
| 1188 ); | |
| 1189 if( (error = alGetError()) != AL_NO_ERROR) | |
| 1190 { | |
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1191 ALmixer_SetError("Can't seek past end and alBufferData failed: %s\n", alGetString(error)); |
| 0 | 1192 return -1; |
| 1193 } | |
| 1194 /* Need to set the loaded_bytes field because I don't trust the OpenAL | |
| 1195 * query command to work because I don't know if it will mutilate the | |
| 1196 * size for its own purposes or return the original size | |
| 1197 */ | |
| 1198 data->loaded_bytes = 8; | |
| 1199 | |
| 1200 /* Not sure if this should be an error or not */ | |
| 1201 /* | |
| 1202 ALmixer_SetError("Can't Seek past end"); | |
| 1203 return -1; | |
| 1204 */ | |
| 1205 return 0; | |
| 1206 } | |
| 1207 | |
| 1208 alBufferData(data->buffer[0], | |
| 1209 TranslateFormat(&data->sample->desired), | |
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1210 &(((ALbyte*)data->sample->buffer)[byte_position]), |
| 0 | 1211 data->total_bytes - byte_position, |
| 1212 data->sample->desired.rate | |
| 1213 ); | |
| 1214 if( (error = alGetError()) != AL_NO_ERROR) | |
| 1215 { | |
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1216 ALmixer_SetError("alBufferData failed: %s\n", alGetString(error)); |
| 0 | 1217 return -1; |
| 1218 } | |
| 1219 /* Need to set the loaded_bytes field because I don't trust the OpenAL | |
| 1220 * query command to work because I don't know if it will mutilate the | |
| 1221 * size for its own purposes or return the original size | |
| 1222 */ | |
| 1223 data->loaded_bytes = data->total_bytes - byte_position; | |
| 1224 | |
| 1225 return 0; | |
| 1226 } | |
| 1227 | |
| 1228 /* Because we have multiple queue buffers and OpenAL won't let | |
| 1229 * us access them, we need to keep copies of each buffer around | |
| 1230 */ | |
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1231 static ALint CopyDataToAccessBuffer(ALmixer_Data* data, ALuint num_bytes, ALuint buffer) |
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1232 { |
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1233 ALint index; |
| 0 | 1234 /* We only want to copy if access_data is true. |
| 1235 * This is determined by whether memory has been | |
| 1236 * allocated in the buffer_map_list or not | |
| 1237 */ | |
| 1238 if(NULL == data->buffer_map_list) | |
| 1239 { | |
| 1240 return -1; | |
| 1241 } | |
| 1242 index = LookUpBuffer(buffer, data->buffer_map_list, data->max_queue_buffers); | |
| 1243 if(-1 == index) | |
| 1244 { | |
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1245 /* |
| 0 | 1246 fprintf(stderr, ">>>>>>>CopyData catch, albuffer=%d\n",buffer); |
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1247 */ |
| 0 | 1248 return -1; |
| 1249 } | |
| 1250 /* Copy the data to the access buffer */ | |
| 1251 memcpy(data->buffer_map_list[index].data, data->sample->buffer, num_bytes); | |
| 1252 data->buffer_map_list[index].num_bytes = data->sample->buffer_size; | |
| 1253 | |
| 1254 return 0; | |
| 1255 } | |
| 1256 | |
| 1257 | |
| 1258 /* For streamed data, gets more data | |
| 1259 * and prepares it in the active Mix_chunk | |
| 1260 */ | |
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1261 static ALuint GetMoreData(ALmixer_Data* data, ALuint buffer) |
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1262 { |
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1263 ALuint bytes_decoded; |
| 0 | 1264 ALenum error; |
| 1265 if(NULL == data) | |
| 1266 { | |
| 1267 ALmixer_SetError("Cannot GetMoreData() because ALmixer_Data* is NULL\n"); | |
| 1268 return 0; | |
| 1269 } | |
| 1270 | |
| 1271 bytes_decoded = Sound_Decode(data->sample); | |
| 1272 if(data->sample->flags & SOUND_SAMPLEFLAG_ERROR) | |
| 1273 { | |
| 1274 fprintf(stderr, "Sound_Decode triggered an ERROR>>>>>>\n"); | |
| 1275 ALmixer_SetError(Sound_GetError()); | |
| 1276 /* Force cleanup through FreeData | |
| 1277 Sound_FreeSample(data->sample); | |
| 1278 */ | |
| 1279 return 0; | |
| 1280 } | |
| 1281 | |
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1282 /* fprintf(stderr, "GetMoreData bytes_decoded=%d\n", bytes_decoded); */ |
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1283 |
| 0 | 1284 /* Don't forget to add check for EOF */ |
| 1285 /* Will return 0 bytes and pass the buck to check sample->flags */ | |
| 1286 if(0 == bytes_decoded) | |
| 1287 { | |
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1288 data->eof = 1; |
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1289 |
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1290 #if 0 |
| 0 | 1291 fprintf(stderr, "Hit eof while trying to buffer\n"); |
| 1292 if(data->sample->flags & SOUND_SAMPLEFLAG_EOF) | |
| 1293 { | |
| 1294 fprintf(stderr, "\tEOF flag\n"); | |
| 1295 } | |
| 1296 if(data->sample->flags & SOUND_SAMPLEFLAG_CANSEEK) | |
| 1297 { | |
| 1298 fprintf(stderr, "\tCanSeek flag\n"); | |
| 1299 } | |
| 1300 if(data->sample->flags & SOUND_SAMPLEFLAG_EAGAIN) | |
| 1301 { | |
| 1302 fprintf(stderr, "\tEAGAIN flag\n"); | |
| 1303 } | |
| 1304 if(data->sample->flags & SOUND_SAMPLEFLAG_NONE) | |
| 1305 { | |
| 1306 fprintf(stderr, "\tNONE flag\n"); | |
| 1307 } | |
| 1308 #endif | |
| 1309 return 0; | |
| 1310 } | |
| 1311 | |
| 1312 #ifdef ENABLE_LOKI_QUEUE_FIX_HACK | |
| 1313 /******* REMOVE ME ********************************/ | |
| 1314 /***************** ANOTHER EXPERIEMENT *******************/ | |
| 1315 /* The PROBLEM: It seems that the Loki distribution has problems | |
| 1316 * with Queuing when the buffer size is not a power of two | |
| 1317 * and additional buffers must come after it. | |
| 1318 * The behavior is inconsistent, but one of several things | |
| 1319 * usually happens: | |
| 1320 * Playback is normal | |
| 1321 * Playback immediately stops after the non-pow2 buffer | |
| 1322 * Playback gets distorted on the non-pow2 buffer | |
| 1323 * The entire program segfaults. | |
| 1324 * The workaround is to always specify a power of two buffer size | |
| 1325 * and hope that SDL_sound always fill it. (By lucky coincidence, | |
| 1326 * I already submitted the Ogg fix.) However, this won't catch | |
| 1327 * cases where a loop happens because the read at the end of the | |
| 1328 * file is typically less than the buffer size. | |
| 1329 * | |
| 1330 * This fix addresses this issue, however it may break in | |
| 1331 * other conditions. Always decode in buffer sizes of powers of 2. | |
| 1332 * | |
| 1333 * The HACK: | |
| 1334 * If the buffer is short, try filling it up with 0's | |
| 1335 * to meet the user requested buffer_size which | |
| 1336 * is probably a nice number OpenAL likes, in | |
| 1337 * hopes to avoid a possible Loki bug with | |
| 1338 * short buffers. If looping (which is the main | |
| 1339 * reason for this), the negative side effect is | |
| 1340 * that it may take longer for the loop to start | |
| 1341 * because it must play dead silence. Or if the decoder | |
| 1342 * doesn't guarantee to return the requested bytes | |
| 1343 * (like Ogg), then you will get breakup in between | |
| 1344 * packets. | |
| 1345 */ | |
| 1346 if( (bytes_decoded) < data->sample->buffer_size) | |
| 1347 { | |
|
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1348 ALubyte bit_depth; |
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1349 ALubyte signedness_value; |
| 0 | 1350 int silence_value; |
| 1351 /* Crap, memset value needs to be the "silent" value, | |
| 1352 * but it will differ for signed/unsigned and bit depth | |
| 1353 */ | |
| 1354 bit_depth = GetBitDepth(data->sample->desired.format); | |
| 1355 signedness_value = GetSignednessValue(data->sample->desired.format); | |
| 1356 if(ALMIXER_SIGNED_VALUE == signedness_value) | |
| 1357 { | |
| 1358 /* I'm guessing that if it's signed, then 0 is the | |
| 1359 * "silent" value */ | |
| 1360 silence_value = 0; | |
| 1361 } | |
| 1362 else | |
| 1363 { | |
| 1364 if(8 == bit_depth) | |
| 1365 { | |
| 1366 /* If 8 bit, I'm guessing it's (2^7)-1 = 127 */ | |
| 1367 silence_value = 127; | |
| 1368 } | |
| 1369 else | |
| 1370 { | |
| 1371 /* For 16 bit, I'm guessing it's (2^15)-1 = 32767 */ | |
| 1372 silence_value = 32767; | |
| 1373 } | |
| 1374 } | |
| 1375 /* Now fill up the rest of the data buffer with the | |
| 1376 * silence_value. | |
| 1377 * I don't think I have to worry about endian issues for | |
| 1378 * this part since the data is for internal use only | |
| 1379 * at this point. | |
| 1380 */ | |
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1381 memset( &( ((ALbyte*)(data->sample->buffer))[bytes_decoded] ), silence_value, data->sample->buffer_size - bytes_decoded); |
| 0 | 1382 /* Now reset the bytes_decoded to reflect the entire |
| 1383 * buffer to tell alBufferData what our full size is. | |
| 1384 */ | |
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1385 /* |
| 0 | 1386 fprintf(stderr, "ALTERED bytes decoded for silence: Original end was %d\n", bytes_decoded); |
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1387 */ |
| 0 | 1388 bytes_decoded = data->sample->buffer_size; |
| 1389 } | |
| 1390 /*********** END EXPERIMENT ******************************/ | |
| 1391 /******* END REMOVE ME ********************************/ | |
| 1392 #endif | |
| 1393 | |
| 1394 /* Now copy the data to the OpenAL buffer */ | |
| 1395 /* We can't just set a pointer because the API needs | |
| 1396 * its own copy to assist hardware acceleration */ | |
| 1397 alBufferData(buffer, | |
| 1398 TranslateFormat(&data->sample->desired), | |
| 1399 data->sample->buffer, | |
| 1400 bytes_decoded, | |
| 1401 data->sample->desired.rate | |
| 1402 ); | |
| 1403 if( (error = alGetError()) != AL_NO_ERROR) | |
| 1404 { | |
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1405 ALmixer_SetError("alBufferData failed: %s\n", alGetString(error)); |
| 0 | 1406 return 0; |
| 1407 } | |
| 1408 | |
| 1409 /* If we need to, copy the data also to the access area | |
| 1410 * (the function will do the check for us) | |
| 1411 */ | |
| 1412 CopyDataToAccessBuffer(data, bytes_decoded, buffer); | |
| 1413 return bytes_decoded; | |
| 1414 } | |
| 1415 | |
| 1416 | |
| 1417 | |
| 1418 | |
| 1419 /******************** EXPERIEMENT **************************** | |
| 1420 * Test function to force maximum buffer filling during loops | |
| 1421 * REMOVE LATER | |
| 1422 *********************************************/ | |
| 1423 #if 0 | |
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1424 static ALint GetMoreData2(ALmixer_Data* data, ALuint buffer) |
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1425 { |
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1426 ALint bytes_decoded; |
| 0 | 1427 ALenum error; |
| 1428 if(NULL == data) | |
| 1429 { | |
| 1430 ALmixer_SetError("Cannot GetMoreData() because ALmixer_Data* is NULL\n"); | |
| 1431 return -1; | |
| 1432 } | |
| 1433 | |
| 1434 if(AL_FALSE == alIsBuffer(buffer)) | |
| 1435 { | |
| 1436 fprintf(stderr, "NOT A BUFFER>>>>>>>>>>>>>>>\n"); | |
| 1437 return -1; | |
| 1438 } | |
| 1439 fprintf(stderr, "Entered GetMoreData222222: buffer id is %d\n", buffer); | |
| 1440 | |
| 1441 /* | |
| 1442 fprintf(stderr, "Decode in GetMoreData\n"); | |
| 1443 */ | |
| 1444 | |
| 1445 #if 0 | |
| 1446 if(buffer%2 == 1) | |
| 1447 { | |
| 1448 fprintf(stderr, "Setting buffer size to 16384\n"); | |
| 1449 Sound_SetBufferSize(data->sample, 16384); | |
| 1450 } | |
| 1451 else | |
| 1452 { | |
| 1453 fprintf(stderr, "Setting buffer size to 8192\n"); | |
| 1454 Sound_SetBufferSize(data->sample, 8192); | |
| 1455 } | |
| 1456 #endif | |
| 1457 | |
| 1458 bytes_decoded = Sound_Decode(data->sample); | |
| 1459 if(data->sample->flags & SOUND_SAMPLEFLAG_ERROR) | |
| 1460 { | |
| 1461 fprintf(stderr, "Sound_Decode triggered an ERROR>>>>>>\n"); | |
| 1462 ALmixer_SetError(Sound_GetError()); | |
| 1463 /* | |
| 1464 Sound_FreeSample(data->sample); | |
| 1465 */ | |
| 1466 return -1; | |
| 1467 } | |
| 1468 /* Don't forget to add check for EOF */ | |
| 1469 /* Will return 0 bytes and pass the buck to check sample->flags */ | |
| 1470 if(0 == bytes_decoded) | |
| 1471 { | |
| 1472 #if 1 | |
| 1473 fprintf(stderr, "Hit eof while trying to buffer\n"); | |
| 1474 data->eof = 1; | |
| 1475 if(data->sample->flags & SOUND_SAMPLEFLAG_EOF) | |
| 1476 { | |
| 1477 fprintf(stderr, "\tEOF flag\n"); | |
| 1478 } | |
| 1479 if(data->sample->flags & SOUND_SAMPLEFLAG_CANSEEK) | |
| 1480 { | |
| 1481 fprintf(stderr, "\tCanSeek flag\n"); | |
| 1482 } | |
| 1483 if(data->sample->flags & SOUND_SAMPLEFLAG_EAGAIN) | |
| 1484 { | |
| 1485 fprintf(stderr, "\tEAGAIN flag\n"); | |
| 1486 } | |
| 1487 if(data->sample->flags & SOUND_SAMPLEFLAG_NONE) | |
| 1488 { | |
| 1489 fprintf(stderr, "\tNONE flag\n"); | |
| 1490 } | |
| 1491 #endif | |
| 1492 return 0; | |
| 1493 } | |
| 1494 | |
| 1495 if(bytes_decoded < 16384) | |
| 1496 { | |
| 1497 char* tempbuffer1 = (char*)malloc(16384); | |
| 1498 char* tempbuffer2 = (char*)malloc(16384); | |
| 1499 int retval; | |
| 1500 memcpy(tempbuffer1, data->sample->buffer, bytes_decoded); | |
| 1501 retval = Sound_SetBufferSize(data->sample, 16384-bytes_decoded); | |
| 1502 if(retval == 1) | |
| 1503 { | |
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1504 ALuint new_bytes; |
| 0 | 1505 Sound_Rewind(data->sample); |
| 1506 new_bytes = Sound_Decode(data->sample); | |
| 1507 fprintf(stderr, "Orig bytes: %d, Make up bytes_decoded=%d, total=%d\n", bytes_decoded, new_bytes, new_bytes+bytes_decoded); | |
| 1508 | |
| 1509 memcpy(tempbuffer2, data->sample->buffer, new_bytes); | |
| 1510 | |
| 1511 retval = Sound_SetBufferSize(data->sample, 16384); | |
| 1512 fprintf(stderr, "Finished reset...now danger copy\n"); | |
| 1513 memcpy(data->sample->buffer, tempbuffer1,bytes_decoded); | |
| 1514 | |
| 1515 fprintf(stderr, "Finished reset...now danger copy2\n"); | |
| 1516 memcpy( &( ((char*)(data->sample->buffer))[bytes_decoded] ), tempbuffer2, new_bytes); | |
| 1517 | |
| 1518 fprintf(stderr, "Finished \n"); | |
| 1519 | |
| 1520 free(tempbuffer1); | |
| 1521 free(tempbuffer2); | |
| 1522 bytes_decoded += new_bytes; | |
| 1523 fprintf(stderr, "ASSERT bytes should equal 16384: %d\n", bytes_decoded); | |
| 1524 } | |
| 1525 else | |
| 1526 { | |
| 1527 fprintf(stderr, "Experiment failed: %s\n", Sound_GetError()); | |
| 1528 } | |
| 1529 } | |
| 1530 | |
| 1531 /* Now copy the data to the OpenAL buffer */ | |
| 1532 /* We can't just set a pointer because the API needs | |
| 1533 * its own copy to assist hardware acceleration */ | |
| 1534 alBufferData(buffer, | |
| 1535 TranslateFormat(&data->sample->desired), | |
| 1536 data->sample->buffer, | |
| 1537 bytes_decoded, | |
| 1538 data->sample->desired.rate | |
| 1539 ); | |
| 1540 if( (error = alGetError()) != AL_NO_ERROR) | |
| 1541 { | |
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1542 ALmixer_SetError("alBufferData failed: %s\n", alGetString(error)); |
| 0 | 1543 return -1; |
| 1544 } | |
| 1545 | |
| 1546 fprintf(stderr, "GetMoreData2222 returning %d bytes decoded\n", bytes_decoded); | |
| 1547 return bytes_decoded; | |
| 1548 } | |
| 1549 #endif | |
| 1550 | |
| 1551 /************ END EXPERIEMENT - REMOVE ME *************************/ | |
| 1552 | |
| 1553 | |
| 1554 | |
| 1555 | |
| 1556 | |
| 1557 | |
| 1558 | |
| 1559 | |
| 1560 | |
| 1561 /* This function will look up the source for the corresponding channel */ | |
| 1562 /* Must return 0 on error instead of -1 because of unsigned int */ | |
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1563 static ALuint Internal_GetSource(ALint channel) |
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1564 { |
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1565 ALint i; |
| 0 | 1566 /* Make sure channel is in bounds */ |
| 1567 if(channel >= Number_of_Channels_global) | |
| 1568 { | |
| 1569 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); | |
| 1570 return 0; | |
| 1571 } | |
| 1572 /* If the user specified -1, then return the an available source */ | |
| 1573 if(channel < 0) | |
| 1574 { | |
| 1575 for(i=Number_of_Reserve_Channels_global; i<Number_of_Channels_global; i++) | |
| 1576 { | |
| 1577 if( ! ALmixer_Channel_List[i].channel_in_use ) | |
| 1578 { | |
| 1579 return ALmixer_Channel_List[i].alsource; | |
| 1580 } | |
| 1581 } | |
| 1582 /* If we get here, all sources are in use */ | |
| 1583 /* Error message seems too harsh | |
| 1584 ALmixer_SetError("All sources are in use"); | |
| 1585 */ | |
| 1586 return 0; | |
| 1587 } | |
| 1588 /* Last case: Return the source for the channel */ | |
| 1589 return ALmixer_Channel_List[channel].alsource; | |
| 1590 } | |
| 1591 | |
| 1592 /* This function will look up the channel for the corresponding source */ | |
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1593 static ALint Internal_GetChannel(ALuint source) |
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1594 { |
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1595 ALint i; |
| 0 | 1596 /* Only the first value is used for the key */ |
| 1597 Source_Map key = { 0, 0 }; | |
| 1598 Source_Map* found_item = NULL; | |
| 1599 key.source = source; | |
| 1600 | |
| 1601 /* If the source is 0, look up the first available channel */ | |
| 1602 if(0 == source) | |
| 1603 { | |
| 1604 for(i=Number_of_Reserve_Channels_global; i<Number_of_Channels_global; i++) | |
| 1605 { | |
| 1606 if( ! ALmixer_Channel_List[i].channel_in_use ) | |
| 1607 { | |
| 1608 return i; | |
| 1609 } | |
| 1610 } | |
| 1611 /* If we get here, all sources are in use */ | |
| 1612 /* Error message seems too harsh | |
| 1613 ALmixer_SetError("All channels are in use"); | |
| 1614 */ | |
| 1615 return -1; | |
| 1616 } | |
| 1617 | |
| 1618 | |
| 1619 /* Else, look up the source and return the channel */ | |
| 1620 if(AL_FALSE == alIsSource(source)) | |
| 1621 { | |
| 1622 ALmixer_SetError("Is not a source"); | |
| 1623 return -1; | |
| 1624 } | |
| 1625 | |
| 1626 /* Use the ANSI C binary search feature (yea!) */ | |
| 1627 found_item = (Source_Map*)bsearch(&key, Source_Map_List, Number_of_Channels_global, sizeof(Source_Map), Compare_Source_Map); | |
| 1628 if(NULL == found_item) | |
| 1629 { | |
| 1630 ALmixer_SetError("Source is valid but not registered with ALmixer (to a channel)"); | |
| 1631 return -1; | |
| 1632 } | |
| 1633 return found_item->channel; | |
| 1634 } | |
| 1635 | |
| 1636 | |
| 1637 | |
| 1638 /* This function will find the first available channel (not in use) | |
| 1639 * from the specified start channel. Reserved channels to not qualify | |
| 1640 * as available. | |
| 1641 */ | |
|
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1642 static ALint Internal_FindFreeChannel(ALint start_channel) |
| 0 | 1643 { |
| 1644 /* Start at the number of reserved so we skip over | |
| 1645 * all the reserved channels. | |
| 1646 */ | |
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1647 ALint i = Number_of_Reserve_Channels_global; |
| 0 | 1648 /* Quick check to see if we're out of bounds */ |
| 1649 if(start_channel >= Number_of_Channels_global) | |
| 1650 { | |
| 1651 return -1; | |
| 1652 } | |
| 1653 | |
| 1654 /* If the start channel is even higher than the reserved, | |
| 1655 * then start at the higher value. | |
| 1656 */ | |
| 1657 if(start_channel > Number_of_Reserve_Channels_global) | |
| 1658 { | |
| 1659 i = start_channel; | |
| 1660 } | |
| 1661 | |
| 1662 /* i has already been set */ | |
| 1663 for( ; i<Number_of_Channels_global; i++) | |
| 1664 { | |
| 1665 if( ! ALmixer_Channel_List[i].channel_in_use ) | |
| 1666 { | |
| 1667 return i; | |
| 1668 } | |
| 1669 } | |
| 1670 /* If we get here, all sources are in use */ | |
| 1671 return -1; | |
| 1672 } | |
| 1673 | |
| 1674 | |
| 1675 | |
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1676 /* Will return the number of channels halted |
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1677 * or 0 for error |
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1678 */ |
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1679 static ALint Internal_HaltChannel(ALint channel, ALboolean did_finish_naturally) |
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1680 { |
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1681 ALint retval = 0; |
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1682 ALint counter = 0; |
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1683 ALenum error; |
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1684 ALint buffers_still_queued; |
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1685 ALint buffers_processed; |
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1686 |
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1687 if(channel >= Number_of_Channels_global) |
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1688 { |
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1689 ALmixer_SetError("Cannot halt channel %d because it exceeds maximum number of channels (%d)\n", channel, Number_of_Channels_global); |
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1690 return -1; |
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1691 } |
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1692 /* If the user specified a specific channel */ |
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1693 if(channel >= 0) |
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1694 { |
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1695 /* only need to process channel if in use */ |
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1696 if(ALmixer_Channel_List[channel].channel_in_use) |
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1697 { |
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1698 alSourceStop(ALmixer_Channel_List[channel].alsource); |
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1699 if((error = alGetError()) != AL_NO_ERROR) |
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1700 { |
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1701 fprintf(stderr, "14Testing error: %s\n", |
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1702 alGetString(error)); |
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1703 } |
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1704 /* Here's the situation. My old method of using |
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1705 * alSourceUnqueueBuffers() seemed to be invalid in light |
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1706 * of all the problems I suffered through with getting |
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1707 * the CoreData backend to work with this code. |
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1708 * As such, I'm changing all the code to set the buffer to |
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1709 * AL_NONE. Furthermore, the queued vs. non-queued issue |
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1710 * doesn't need to apply here. For non-queued, Loki, |
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1711 * Creative Windows, and CoreAudio seem to leave the |
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1712 * buffer queued (Old Mac didn't.) For queued, we need to |
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1713 * remove the processed buffers and force remove the |
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1714 * still-queued buffers. |
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1715 */ |
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1716 alGetSourcei( |
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1717 ALmixer_Channel_List[channel].alsource, |
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1718 AL_BUFFERS_QUEUED, &buffers_still_queued |
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1719 ); |
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1720 if((error = alGetError()) != AL_NO_ERROR) |
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1721 { |
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1722 fprintf(stderr, "17Testing Error with buffers_still_queued: %s", |
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1723 alGetString(error)); |
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1724 ALmixer_SetError("Failed detecting still queued buffers: %s", |
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1725 alGetString(error) ); |
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1726 retval = -1; |
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1727 } |
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1728 alGetSourcei( |
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1729 ALmixer_Channel_List[channel].alsource, |
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1730 AL_BUFFERS_PROCESSED, &buffers_processed |
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1731 ); |
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1732 if((error = alGetError()) != AL_NO_ERROR) |
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1733 { |
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1734 fprintf(stderr, "17Testing Error with buffers_processed: %s", |
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1735 alGetString(error)); |
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1736 ALmixer_SetError("Failed detecting still processed buffers: %s", |
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1737 alGetString(error) ); |
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1738 retval = -1; |
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1739 } |
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1740 /* If either of these is greater than 0, it means we need |
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1741 * to clear the source |
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1742 */ |
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1743 if((buffers_still_queued > 0) || (buffers_processed > 0)) |
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1744 { |
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1745 alSourcei(ALmixer_Channel_List[channel].alsource, |
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1746 AL_BUFFER, |
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1747 AL_NONE); |
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1748 if((error = alGetError()) != AL_NO_ERROR) |
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1749 { |
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1750 fprintf(stderr, "17Testing Error with clearing buffer from source: %s", |
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1751 alGetString(error)); |
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1752 ALmixer_SetError("Failed to clear buffer from source: %s", |
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1753 alGetString(error) ); |
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1754 retval = -1; |
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1755 } |
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1756 } |
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1757 |
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1758 ALmixer_Channel_List[channel].almixer_data->num_buffers_in_use = 0; |
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1759 |
|
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1760 /* Launch callback for consistency? */ |
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1761 Invoke_Channel_Done_Callback(channel, did_finish_naturally); |
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1762 |
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1763 Clean_Channel(channel); |
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1764 Is_Playing_global--; |
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1765 counter++; |
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1766 } |
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1767 } |
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1768 /* The user wants to halt all channels */ |
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1769 else |
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1770 { |
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1771 ALint i; |
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1772 for(i=0; i<Number_of_Channels_global; i++) |
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1773 { |
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1774 /* only need to process channel if in use */ |
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1775 if(ALmixer_Channel_List[i].channel_in_use) |
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1776 { |
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1777 alSourceStop(ALmixer_Channel_List[i].alsource); |
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1778 if((error = alGetError()) != AL_NO_ERROR) |
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1779 { |
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1780 fprintf(stderr, "19Testing error: %s\n", |
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1781 alGetString(error)); |
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1782 } |
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1783 |
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1784 /* Here's the situation. My old method of using |
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1785 * alSourceUnqueueBuffers() seemed to be invalid in light |
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1786 * of all the problems I suffered through with getting |
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1787 * the CoreData backend to work with this code. |
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1788 * As such, I'm changing all the code to set the buffer to |
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1789 * AL_NONE. Furthermore, the queued vs. non-queued issue |
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1790 * doesn't need to apply here. For non-queued, Loki, |
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1791 * Creative Windows, and CoreAudio seem to leave the |
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1792 * buffer queued (Old Mac didn't.) For queued, we need to |
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1793 * remove the processed buffers and force remove the |
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1794 * still-queued buffers. |
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1795 */ |
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1796 alGetSourcei( |
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1797 ALmixer_Channel_List[i].alsource, |
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1798 AL_BUFFERS_QUEUED, &buffers_still_queued |
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1799 ); |
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1800 if((error = alGetError()) != AL_NO_ERROR) |
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1801 { |
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1802 fprintf(stderr, "17Testing Error with buffers_still_queued: %s", |
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1803 alGetString(error)); |
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1804 ALmixer_SetError("Failed detecting still queued buffers: %s", |
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1805 alGetString(error) ); |
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1806 retval = -1; |
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1807 } |
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1808 alGetSourcei( |
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1809 ALmixer_Channel_List[i].alsource, |
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1810 AL_BUFFERS_PROCESSED, &buffers_processed |
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1811 ); |
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1812 if((error = alGetError()) != AL_NO_ERROR) |
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1813 { |
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1814 fprintf(stderr, "17Testing Error with buffers_processed: %s", |
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1815 alGetString(error)); |
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1816 ALmixer_SetError("Failed detecting still processed buffers: %s", |
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1817 alGetString(error) ); |
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1818 retval = -1; |
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1819 } |
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1820 /* If either of these is greater than 0, it means we need |
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1821 * to clear the source |
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1822 */ |
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1823 if((buffers_still_queued > 0) || (buffers_processed > 0)) |
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1824 { |
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1825 alSourcei(ALmixer_Channel_List[i].alsource, |
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1826 AL_BUFFER, |
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1827 AL_NONE); |
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1828 if((error = alGetError()) != AL_NO_ERROR) |
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1829 { |
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1830 fprintf(stderr, "17Testing Error with clearing buffer from source: %s", |
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1831 alGetString(error)); |
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1832 ALmixer_SetError("Failed to clear buffer from source: %s", |
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1833 alGetString(error) ); |
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1834 retval = -1; |
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1835 } |
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1836 } |
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1837 |
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1838 ALmixer_Channel_List[i].almixer_data->num_buffers_in_use = 0; |
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1839 |
|
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1840 /* Launch callback for consistency? */ |
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1841 Invoke_Channel_Done_Callback(i, did_finish_naturally); |
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1842 |
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1843 Clean_Channel(i); |
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1844 Is_Playing_global--; |
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1845 |
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1846 /* Increment the counter */ |
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1847 counter++; |
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1848 } |
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1849 /* Let's halt everything just in case there |
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1850 * are bugs. |
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1851 */ |
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1852 /* |
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1853 else |
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1854 { |
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1855 alSourceStop(ALmixer_Channel_List[channel].alsource); |
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1856 / * Can't clean because the in_use counter for |
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1857 * data will get messed up * / |
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1858 Clean_Channel(channel); |
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1859 } |
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1860 */ |
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1861 /* Just in case */ |
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1862 Is_Playing_global = 0; |
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1863 } |
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1864 } |
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1865 if(-1 == retval) |
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1866 { |
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1867 return -1; |
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1868 } |
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1869 return counter; |
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1870 } |
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1871 |
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1872 |
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1873 /* Will return the source halted or the total number of channels |
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1874 * if all were halted or 0 for error |
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1875 */ |
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1876 static ALint Internal_HaltSource(ALuint source, ALboolean did_finish_naturally) |
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1877 { |
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1878 ALint channel; |
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1879 if(0 == source) |
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1880 { |
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1881 /* Will return the number of sources halted */ |
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1882 return Internal_HaltChannel(-1, did_finish_naturally); |
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1883 } |
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1884 |
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1885 channel = Internal_GetChannel(source); |
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1886 if(-1 == channel) |
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1887 { |
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1888 ALmixer_SetError("Cannot halt source: %s", ALmixer_GetError()); |
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1889 return -1; |
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1890 } |
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1891 return Internal_HaltChannel(channel, did_finish_naturally); |
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1892 } |
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1893 |
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1894 |
| 0 | 1895 |
| 1896 /* Note: Behaves, almost like SDL_mixer, but keep in mind | |
| 1897 * that there is no "music" channel anymore, so 0 | |
| 1898 * will remove everything. (Note, I no longer allow 0 | |
| 1899 * so it gets set to the default number.) | |
| 1900 * Also, callbacks for deleted channels will not be called. | |
| 1901 * I really need to do error checking, for realloc and | |
| 1902 * GenSources, but reversing the damage is too painful | |
| 1903 * for me to think about at the moment, so it's not in here. | |
| 1904 */ | |
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1905 static ALint Internal_AllocateChannels(ALint numchans) |
| 0 | 1906 { |
| 1907 ALenum error; | |
| 1908 int i; | |
| 1909 /* Return info */ | |
| 1910 if(numchans < 0) | |
| 1911 { | |
| 1912 return Number_of_Channels_global; | |
| 1913 } | |
| 1914 if(0 == numchans) | |
| 1915 { | |
| 1916 numchans = ALMIXER_DEFAULT_NUM_CHANNELS; | |
| 1917 } | |
| 1918 /* No change */ | |
| 1919 if(numchans == Number_of_Channels_global) | |
| 1920 { | |
| 1921 return Number_of_Channels_global; | |
| 1922 } | |
| 1923 /* We need to increase the number of channels */ | |
| 1924 if(numchans > Number_of_Channels_global) | |
| 1925 { | |
| 1926 /* Not sure how safe this is, but SDL_mixer does it | |
| 1927 * the same way */ | |
| 1928 ALmixer_Channel_List = (struct ALmixer_Channel*) realloc( ALmixer_Channel_List, numchans * sizeof(struct ALmixer_Channel)); | |
| 1929 | |
| 1930 /* Allocate memory for the list of sources that map to the channels */ | |
| 1931 Source_Map_List = (Source_Map*) realloc(Source_Map_List, numchans * sizeof(Source_Map)); | |
| 1932 | |
| 1933 for(i=Number_of_Channels_global; i<numchans; i++) | |
| 1934 { | |
| 1935 Init_Channel(i); | |
| 1936 /* Generate a new source and associate it with the channel */ | |
| 1937 alGenSources(1, &ALmixer_Channel_List[i].alsource); | |
| 1938 if((error = alGetError()) != AL_NO_ERROR) | |
| 1939 { | |
| 1940 fprintf(stderr, "12Testing errpr before unqueue because getting stuff, for OS X this is expected: %s\n", | |
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1941 alGetString(error)); |
| 0 | 1942 } |
| 1943 /* Copy the source so the SourceMap has it too */ | |
| 1944 Source_Map_List[i].source = ALmixer_Channel_List[i].alsource; | |
| 1945 Source_Map_List[i].channel = i; | |
| 1946 /* Clean the channel because there are some things that need to | |
| 1947 * be done that can't happen until the source is set | |
| 1948 */ | |
| 1949 Clean_Channel(i); | |
| 1950 } | |
| 1951 | |
| 1952 /* The Source_Map_List must be sorted by source for binary searches | |
| 1953 */ | |
| 1954 qsort(Source_Map_List, numchans, sizeof(Source_Map), Compare_Source_Map); | |
| 1955 | |
| 1956 Number_of_Channels_global = numchans; | |
| 1957 return numchans; | |
| 1958 } | |
| 1959 /* Need to remove channels. This might be dangerous */ | |
| 1960 if(numchans < Number_of_Channels_global) | |
| 1961 { | |
| 1962 for(i=numchans; i<Number_of_Channels_global; i++) | |
| 1963 { | |
| 1964 /* Halt the channel */ | |
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1965 Internal_HaltChannel(i, AL_FALSE); |
| 0 | 1966 |
| 1967 /* Delete source associated with the channel */ | |
| 1968 alDeleteSources(1, &ALmixer_Channel_List[i].alsource); | |
| 1969 if((error = alGetError()) != AL_NO_ERROR) | |
| 1970 { | |
| 1971 fprintf(stderr, "13Testing error: %s\n", | |
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1972 alGetString(error)); |
| 0 | 1973 } |
| 1974 } | |
| 1975 | |
| 1976 | |
| 1977 /* Not sure how safe this is, but SDL_mixer does it | |
| 1978 * the same way */ | |
| 1979 ALmixer_Channel_List = (struct ALmixer_Channel*) realloc( ALmixer_Channel_List, numchans * sizeof(struct ALmixer_Channel)); | |
| 1980 | |
| 1981 /* The tricky part is that we must remove the entries | |
| 1982 * in the source map that correspond to the deleted channels. | |
| 1983 * We'll resort the map by channels so we can pick them off | |
| 1984 * in order. | |
| 1985 */ | |
| 1986 qsort(Source_Map_List, Number_of_Channels_global, sizeof(Source_Map), Compare_Source_Map_by_channel); | |
| 1987 | |
| 1988 /* Deallocate memory for the list of sources that map to the channels */ | |
| 1989 Source_Map_List = (Source_Map*) realloc(Source_Map_List, numchans * sizeof(Source_Map)); | |
| 1990 | |
| 1991 /* Now resort the map by source and the correct num of chans */ | |
| 1992 qsort(Source_Map_List, numchans, sizeof(Source_Map), Compare_Source_Map); | |
| 1993 | |
| 1994 /* Reset the number of channels */ | |
| 1995 Number_of_Channels_global = numchans; | |
| 1996 return numchans; | |
| 1997 } | |
| 1998 /* Shouldn't ever reach here */ | |
| 1999 return -1; | |
| 2000 | |
| 2001 } | |
| 2002 | |
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2003 static ALint Internal_ReserveChannels(ALint num) |
| 0 | 2004 { |
| 2005 /* Can't reserve more than the max num of channels */ | |
| 2006 /* Actually, I'll allow it for people who just want to | |
| 2007 * set the value really high to effectively disable | |
| 2008 * auto-assignment | |
| 2009 */ | |
| 2010 | |
| 2011 /* Return the current number of reserved channels */ | |
| 2012 if(num < 0) | |
| 2013 { | |
| 2014 return Number_of_Reserve_Channels_global; | |
| 2015 } | |
| 2016 Number_of_Reserve_Channels_global = num; | |
| 2017 return Number_of_Reserve_Channels_global; | |
| 2018 } | |
| 2019 | |
| 2020 | |
| 2021 /* This will rewind the SDL_Sound sample for streamed | |
| 2022 * samples and start buffering up the data for the next | |
| 2023 * playback. This may require samples to be halted | |
| 2024 */ | |
|
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2025 static ALboolean Internal_RewindData(ALmixer_Data* data) |
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2026 { |
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2027 ALint retval = 0; |
| 0 | 2028 /* |
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2029 ALint bytes_returned; |
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2030 ALint i; |
| 0 | 2031 */ |
| 2032 if(NULL == data) | |
| 2033 { | |
| 2034 ALmixer_SetError("Cannot rewind because data is NULL\n"); | |
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2035 return AL_FALSE; |
| 0 | 2036 } |
| 2037 | |
| 2038 | |
| 2039 /* Might have to require Halt */ | |
| 2040 /* Okay, we assume Halt or natural stop has already | |
| 2041 * cleared the data buffers | |
| 2042 */ | |
| 2043 if(data->in_use) | |
| 2044 { | |
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2045 /* |
| 0 | 2046 fprintf(stderr, "Warning sample is in use. May not be able to rewind\n"); |
|
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|
2047 */ |
| 0 | 2048 /* |
| 2049 ALmixer_SetError("Data is in use. Cannot rewind unless all sources using the data are halted\n"); | |
| 2050 return -1; | |
| 2051 */ | |
| 2052 } | |
| 2053 | |
| 2054 | |
| 2055 /* Because Seek can alter things even in predecoded data, | |
| 2056 * decoded data must also be rewound | |
| 2057 */ | |
| 2058 if(data->decoded_all) | |
| 2059 { | |
| 2060 data->eof = 0; | |
| 2061 | |
| 2062 #if 0 | |
| 2063 #if defined(DISABLE_PREDECODED_SEEK) | |
| 2064 /* Since we can't seek predecoded stuff, it should be rewound */ | |
|
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2065 return AL_TRUE; |
| 0 | 2066 #elif !defined(DISABLE_SEEK_MEMORY_OPTIMIZATION) |
| 2067 /* This case is if the Sound_Sample has been deleted. | |
| 2068 * It assumes the data is already at the beginning. | |
| 2069 */ | |
| 2070 if(NULL == data->sample) | |
| 2071 { | |
|
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2072 return AL_TRUE; |
| 0 | 2073 } |
| 2074 /* Else, the sample has already been reallocated, | |
| 2075 * and we can fall to normal behavior | |
| 2076 */ | |
| 2077 #endif | |
| 2078 #endif | |
| 2079 /* If access_data, was enabled, the sound sample | |
| 2080 * still exists and we can do stuff. | |
| 2081 * If it's NULL, we can't do anything, but | |
| 2082 * it should already be "rewound". | |
| 2083 */ | |
| 2084 if(NULL == data->sample) | |
| 2085 { | |
|
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2086 return AL_TRUE; |
| 0 | 2087 } |
| 2088 /* Else, the sample has already been reallocated, | |
| 2089 * and we can fall to normal behavior | |
| 2090 */ | |
| 2091 | |
| 2092 Set_Predecoded_Seek_Position(data, 0); | |
| 2093 /* | |
| 2094 return data->total_bytes; | |
| 2095 */ | |
|
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2096 return AL_TRUE; |
| 0 | 2097 } |
| 2098 | |
| 2099 /* Remaining stuff for streamed data */ | |
| 2100 | |
| 2101 data->eof = 0; | |
| 2102 retval = Sound_Rewind(data->sample); | |
| 2103 if(0 == retval) | |
| 2104 { | |
| 2105 ALmixer_SetError( Sound_GetError() ); | |
|
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2106 return AL_FALSE; |
| 0 | 2107 } |
| 2108 #if 0 | |
| 2109 /* Clear error */ | |
| 2110 alGetError(); | |
| 2111 for(i=0; i<data->num_buffers; i++) | |
| 2112 { | |
| 2113 bytes_returned = GetMoreData(data, data->buffer[i]); | |
| 2114 if(-1 == bytes_returned) | |
| 2115 { | |
|
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2116 return AL_FALSE; |
| 0 | 2117 } |
| 2118 else if(0 == bytes_returned) | |
| 2119 { | |
|
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2120 return AL_FALSE; |
| 0 | 2121 } |
| 2122 retval += bytes_returned; | |
| 2123 | |
| 2124 } | |
| 2125 #endif | |
| 2126 | |
| 2127 | |
| 2128 | |
|
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2129 return AL_TRUE; |
| 0 | 2130 } |
| 2131 | |
| 2132 | |
| 2133 | |
| 2134 | |
|
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2135 static ALint Internal_RewindChannel(ALint channel) |
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2136 { |
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2137 ALint retval = 0; |
| 0 | 2138 ALenum error; |
| 2139 ALint state; | |
|
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2140 ALint running_count = 0; |
| 0 | 2141 |
| 2142 if(channel >= Number_of_Channels_global) | |
| 2143 { | |
| 2144 ALmixer_SetError("Cannot rewind channel %d because it exceeds maximum number of channels (%d)\n", channel, Number_of_Channels_global); | |
| 2145 return -1; | |
| 2146 } | |
| 2147 | |
| 2148 if((error = alGetError()) != AL_NO_ERROR) | |
| 2149 { | |
| 2150 fprintf(stderr, "24Testing error: %s\n", | |
|
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2151 alGetString(error)); |
| 0 | 2152 } |
| 2153 /* Clear error */ | |
| 2154 alGetError(); | |
| 2155 | |
| 2156 /* If the user specified a specific channel */ | |
| 2157 if(channel >= 0) | |
| 2158 { | |
| 2159 /* only need to process channel if in use */ | |
| 2160 if(ALmixer_Channel_List[channel].channel_in_use) | |
| 2161 { | |
| 2162 | |
| 2163 /* What should I do? Do I just rewind the channel | |
| 2164 * or also rewind the data? Since the data is | |
| 2165 * shared, let's make it the user's responsibility | |
| 2166 * to rewind the data. | |
| 2167 */ | |
| 2168 if(ALmixer_Channel_List[channel].almixer_data->decoded_all) | |
| 2169 { | |
| 2170 alGetSourcei( | |
| 2171 ALmixer_Channel_List[channel].alsource, | |
| 2172 AL_SOURCE_STATE, &state | |
| 2173 ); | |
| 2174 if((error = alGetError()) != AL_NO_ERROR) | |
| 2175 { | |
| 2176 fprintf(stderr, "25Testing error: %s\n", | |
|
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2177 alGetString(error)); |
| 0 | 2178 } |
| 2179 alSourceRewind(ALmixer_Channel_List[channel].alsource); | |
| 2180 if((error = alGetError()) != AL_NO_ERROR) | |
| 2181 { | |
| 2182 ALmixer_SetError("%s", | |
|
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2183 alGetString(error) ); |
| 0 | 2184 retval = -1; |
| 2185 } | |
| 2186 /* Need to resume playback if it was originally playing */ | |
| 2187 if(AL_PLAYING == state) | |
| 2188 { | |
| 2189 alSourcePlay(ALmixer_Channel_List[channel].alsource); | |
| 2190 if((error = alGetError()) != AL_NO_ERROR) | |
| 2191 { | |
| 2192 ALmixer_SetError("%s", | |
|
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2193 alGetString(error) ); |
| 0 | 2194 retval = -1; |
| 2195 } | |
| 2196 } | |
| 2197 else if(AL_PAUSED == state) | |
| 2198 { | |
| 2199 /* HACK: The problem is that when paused, after | |
| 2200 * the Rewind, I can't get it off the INITIAL | |
| 2201 * state without restarting | |
| 2202 */ | |
| 2203 alSourcePlay(ALmixer_Channel_List[channel].alsource); | |
| 2204 if((error = alGetError()) != AL_NO_ERROR) | |
| 2205 { | |
| 2206 fprintf(stderr, "25Testing error: %s\n", | |
|
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2207 alGetString(error)); |
| 0 | 2208 } |
| 2209 alSourcePause(ALmixer_Channel_List[channel].alsource); | |
| 2210 if((error = alGetError()) != AL_NO_ERROR) | |
| 2211 { | |
| 2212 ALmixer_SetError("%s", | |
|
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2213 alGetString(error) ); |
| 0 | 2214 retval = -1; |
| 2215 } | |
| 2216 } | |
| 2217 } | |
| 2218 else | |
| 2219 { | |
| 2220 /* Streamed data is different. Rewinding the channel | |
| 2221 * does no good. Rewinding the data will have an | |
| 2222 * effect, but it will be lagged based on how | |
| 2223 * much data is queued. Recommend users call Halt | |
| 2224 * before rewind if they want immediate results. | |
| 2225 */ | |
| 2226 retval = Internal_RewindData(ALmixer_Channel_List[channel].almixer_data); | |
| 2227 } | |
| 2228 } | |
| 2229 } | |
|
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2230 /* The user wants to rewind all channels */ |
| 0 | 2231 else |
| 2232 { | |
|
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2233 ALint i; |
| 0 | 2234 for(i=0; i<Number_of_Channels_global; i++) |
| 2235 { | |
| 2236 /* only need to process channel if in use */ | |
| 2237 if(ALmixer_Channel_List[i].channel_in_use) | |
| 2238 { | |
| 2239 /* What should I do? Do I just rewind the channel | |
| 2240 * or also rewind the data? Since the data is | |
| 2241 * shared, let's make it the user's responsibility | |
| 2242 * to rewind the data. | |
| 2243 */ | |
| 2244 if(ALmixer_Channel_List[i].almixer_data->decoded_all) | |
| 2245 { | |
| 2246 alGetSourcei( | |
| 2247 ALmixer_Channel_List[i].alsource, | |
| 2248 AL_SOURCE_STATE, &state | |
| 2249 ); | |
| 2250 if((error = alGetError()) != AL_NO_ERROR) | |
| 2251 { | |
| 2252 fprintf(stderr, "26Testing error: %s\n", | |
|
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2253 alGetString(error)); |
| 0 | 2254 } |
| 2255 alSourceRewind(ALmixer_Channel_List[i].alsource); | |
| 2256 if((error = alGetError()) != AL_NO_ERROR) | |
| 2257 { | |
| 2258 ALmixer_SetError("%s", | |
|
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2259 alGetString(error) ); |
| 0 | 2260 retval = -1; |
| 2261 } | |
| 2262 /* Need to resume playback if it was originally playing */ | |
| 2263 if(AL_PLAYING == state) | |
| 2264 { | |
| 2265 alSourcePlay(ALmixer_Channel_List[i].alsource); | |
| 2266 if((error = alGetError()) != AL_NO_ERROR) | |
| 2267 { | |
| 2268 ALmixer_SetError("%s", | |
|
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2269 alGetString(error) ); |
| 0 | 2270 retval = -1; |
| 2271 } | |
| 2272 } | |
| 2273 else if(AL_PAUSED == state) | |
| 2274 { | |
| 2275 /* HACK: The problem is that when paused, after | |
| 2276 * the Rewind, I can't get it off the INITIAL | |
| 2277 * state without restarting | |
| 2278 */ | |
| 2279 alSourcePlay(ALmixer_Channel_List[i].alsource); | |
| 2280 if((error = alGetError()) != AL_NO_ERROR) | |
| 2281 { | |
| 2282 fprintf(stderr, "27Testing error: %s\n", | |
|
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2283 alGetString(error)); |
| 0 | 2284 } |
| 2285 alSourcePause(ALmixer_Channel_List[i].alsource); | |
| 2286 if((error = alGetError()) != AL_NO_ERROR) | |
| 2287 { | |
| 2288 ALmixer_SetError("%s", | |
|
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2289 alGetString(error) ); |
| 0 | 2290 retval = -1; |
| 2291 } | |
| 2292 } | |
| 2293 } | |
| 2294 else | |
| 2295 { | |
| 2296 /* Streamed data is different. Rewinding the channel | |
| 2297 * does no good. Rewinding the data will have an | |
| 2298 * effect, but it will be lagged based on how | |
| 2299 * much data is queued. Recommend users call Halt | |
| 2300 * before rewind if they want immediate results. | |
| 2301 */ | |
|
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2302 running_count += Internal_RewindData(ALmixer_Channel_List[i].almixer_data); |
| 0 | 2303 } |
| 2304 } | |
| 2305 } | |
| 2306 } | |
|
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2307 if(-1 == retval) |
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2308 { |
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2309 return -1; |
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2310 } |
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2311 else |
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2312 { |
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2313 return running_count; |
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2314 } |
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2315 |
| 0 | 2316 } |
| 2317 | |
| 2318 | |
|
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2319 static ALint Internal_RewindSource(ALuint source) |
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2320 { |
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2321 ALint channel; |
| 0 | 2322 if(0 == source) |
| 2323 { | |
|
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2324 return Internal_RewindChannel(-1) + 1; |
| 0 | 2325 } |
| 2326 | |
| 2327 channel = Internal_GetChannel(source); | |
| 2328 if(-1 == channel) | |
| 2329 { | |
| 2330 ALmixer_SetError("Cannot rewind source: %s", ALmixer_GetError()); | |
| 2331 return 0; | |
| 2332 } | |
|
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2333 return Internal_RewindChannel(channel) + 1; |
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2334 } |
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2335 |
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2336 |
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2337 |
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2338 |
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2339 |
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2340 static ALint Internal_PlayChannelTimed(ALint channel, ALmixer_Data* data, ALint loops, ALint ticks) |
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2341 { |
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2342 ALenum error; |
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2343 int ret_flag = 0; |
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2344 if(NULL == data) |
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2345 { |
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|
2346 ALmixer_SetError("Can't play because data is NULL\n"); |
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2347 return -1; |
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2348 } |
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2349 |
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2350 /* There isn't a good way to share streamed files because |
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2351 * the decoded data doesn't stick around. |
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2352 * You must "Load" a brand new instance of |
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2353 * the data. If you try using the same data, |
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2354 * bad things may happen. This check will attempt |
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2355 * to prevent sharing |
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2356 */ |
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2357 if(0 == data->decoded_all) |
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2358 { |
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2359 if(data->in_use) |
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2360 { |
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|
2361 ALmixer_SetError("Can't play shared streamed sample because it is already in use"); |
|
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diff
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|
2362 return -1; |
|
279d0427ef26
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parents:
1
diff
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|
2363 } |
|
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parents:
1
diff
changeset
|
2364 |
|
279d0427ef26
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parents:
1
diff
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|
2365 /* Make sure SDL_sound sample is not at EOF. |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
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diff
changeset
|
2366 * This mainly affects streamed files, |
|
279d0427ef26
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parents:
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diff
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|
2367 * so the check is placed here |
|
279d0427ef26
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parents:
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diff
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|
2368 */ |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
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|
2369 if(data->eof) |
|
279d0427ef26
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parents:
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diff
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|
2370 { |
|
279d0427ef26
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parents:
1
diff
changeset
|
2371 if( -1 == Internal_RewindData(data) ) |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
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|
2372 { |
|
279d0427ef26
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parents:
1
diff
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|
2373 ALmixer_SetError("Can't play sample because it is at EOF and cannot rewind"); |
|
279d0427ef26
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1
diff
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|
2374 return -1; |
|
279d0427ef26
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parents:
1
diff
changeset
|
2375 } |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
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|
2376 } |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
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|
2377 } |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
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|
2378 /* We need to provide the user with the first available channel */ |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
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|
2379 if(-1 == channel) |
|
279d0427ef26
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diff
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|
2380 { |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
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1
diff
changeset
|
2381 ALint i; |
|
279d0427ef26
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diff
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|
2382 for(i=Number_of_Reserve_Channels_global; i<Number_of_Channels_global; i++) |
|
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diff
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|
2383 { |
|
279d0427ef26
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parents:
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diff
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|
2384 if(0 == ALmixer_Channel_List[i].channel_in_use) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
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diff
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|
2385 { |
|
279d0427ef26
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1
diff
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|
2386 channel = i; |
|
279d0427ef26
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parents:
1
diff
changeset
|
2387 break; |
|
279d0427ef26
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parents:
1
diff
changeset
|
2388 } |
|
279d0427ef26
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parents:
1
diff
changeset
|
2389 } |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
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|
2390 /* if we couldn't find a channel, return an error */ |
|
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1
diff
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|
2391 if(i == Number_of_Channels_global) |
|
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diff
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|
2392 { |
|
279d0427ef26
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diff
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|
2393 ALmixer_SetError("No channels available for playing"); |
|
279d0427ef26
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diff
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|
2394 return -1; |
|
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parents:
1
diff
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|
2395 } |
|
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parents:
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diff
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|
2396 } |
|
279d0427ef26
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parents:
1
diff
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|
2397 /* If we didn't assign the channel number, make sure it's not |
|
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diff
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|
2398 * out of bounds or in use */ |
|
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1
diff
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|
2399 else |
|
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diff
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|
2400 { |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
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diff
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|
2401 if(channel >= Number_of_Channels_global) |
|
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1
diff
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|
2402 { |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
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|
2403 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
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1
diff
changeset
|
2404 return -1; |
|
279d0427ef26
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diff
changeset
|
2405 } |
|
279d0427ef26
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parents:
1
diff
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|
2406 else if(ALmixer_Channel_List[channel].channel_in_use) |
|
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1
diff
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|
2407 { |
|
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1
diff
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|
2408 ALmixer_SetError("Requested channel (%d) is in use", channel, Number_of_Channels_global-1, Number_of_Channels_global); |
|
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diff
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|
2409 return -1; |
|
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1
diff
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|
2410 } |
|
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1
diff
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|
2411 } |
|
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parents:
1
diff
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|
2412 /* Make sure the user doesn't enter some meaningless value */ |
|
279d0427ef26
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parents:
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diff
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|
2413 if(loops < -1) |
|
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1
diff
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|
2414 { |
|
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parents:
1
diff
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|
2415 loops = -1; |
|
279d0427ef26
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1
diff
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|
2416 } |
|
279d0427ef26
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parents:
1
diff
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|
2417 |
|
279d0427ef26
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1
diff
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|
2418 /* loops will probably have to change to be controlled by SDL_Sound */ |
|
279d0427ef26
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1
diff
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|
2419 |
|
279d0427ef26
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1
diff
changeset
|
2420 /* Set up the initial values for playing */ |
|
279d0427ef26
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1
diff
changeset
|
2421 ALmixer_Channel_List[channel].channel_in_use = 1; |
|
279d0427ef26
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diff
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|
2422 data->in_use++; |
|
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diff
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|
2423 |
|
279d0427ef26
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1
diff
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|
2424 /* Shouldn't need updating until a callback is fired |
|
279d0427ef26
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1
diff
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|
2425 * (assuming that we call Play in this function |
|
279d0427ef26
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1
diff
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|
2426 */ |
|
279d0427ef26
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1
diff
changeset
|
2427 ALmixer_Channel_List[channel].needs_stream = 0; |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
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1
diff
changeset
|
2428 ALmixer_Channel_List[channel].almixer_data = data; |
|
279d0427ef26
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parents:
1
diff
changeset
|
2429 ALmixer_Channel_List[channel].start_time = ALmixer_GetTicks(); |
|
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diff
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|
2430 |
|
279d0427ef26
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parents:
1
diff
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|
2431 /* If user entered -1 (or less), set to -1 */ |
|
279d0427ef26
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parents:
1
diff
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|
2432 if(ticks < 0) |
|
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1
diff
changeset
|
2433 { |
|
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
2434 ALmixer_Channel_List[channel].expire_ticks = -1; |
|
279d0427ef26
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1
diff
changeset
|
2435 } |
|
279d0427ef26
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parents:
1
diff
changeset
|
2436 else |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
2437 { |
|
279d0427ef26
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1
diff
changeset
|
2438 ALmixer_Channel_List[channel].expire_ticks = ticks; |
|
279d0427ef26
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diff
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|
2439 } |
|
279d0427ef26
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parents:
1
diff
changeset
|
2440 |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
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|
2441 |
|
279d0427ef26
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1
diff
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|
2442 ALmixer_Channel_List[channel].halted = 0; |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
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1
diff
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|
2443 ALmixer_Channel_List[channel].paused = 0; |
|
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1
diff
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|
2444 |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
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1
diff
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|
2445 /* Ran just use OpenAL to control loops if predecoded and infinite */ |
|
279d0427ef26
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1
diff
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|
2446 ALmixer_Channel_List[channel].loops = loops; |
|
279d0427ef26
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1
diff
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|
2447 if( (-1 == loops) && (data->decoded_all) ) |
|
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1
diff
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|
2448 { |
|
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1
diff
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|
2449 alSourcei(ALmixer_Channel_List[channel].alsource, AL_LOOPING, AL_TRUE); |
|
279d0427ef26
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|
2450 } |
|
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diff
changeset
|
2451 else |
|
279d0427ef26
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1
diff
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|
2452 { |
|
279d0427ef26
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1
diff
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|
2453 alSourcei(ALmixer_Channel_List[channel].alsource, AL_LOOPING, AL_FALSE); |
|
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diff
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|
2454 } |
|
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1
diff
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|
2455 if((error = alGetError()) != AL_NO_ERROR) |
|
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1
diff
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|
2456 { |
|
279d0427ef26
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1
diff
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|
2457 fprintf(stderr, "13Testing error: %s\n", |
|
279d0427ef26
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1
diff
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|
2458 alGetString(error)); |
|
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diff
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|
2459 } |
|
279d0427ef26
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1
diff
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|
2460 |
|
279d0427ef26
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parents:
1
diff
changeset
|
2461 #if 0 |
|
279d0427ef26
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1
diff
changeset
|
2462 /* Because of the corner case, predecoded |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
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1
diff
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|
2463 * files must add +1 to the loops. |
|
279d0427ef26
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1
diff
changeset
|
2464 * Streams do not have this problem |
|
279d0427ef26
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1
diff
changeset
|
2465 * because they can use the eof flag to |
|
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1
diff
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|
2466 * avoid the conflict. |
|
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1
diff
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|
2467 * Sharing data chunks prevents the use of the eof flag. |
|
279d0427ef26
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1
diff
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|
2468 * Since streams, cannot share, only predecoded |
|
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1
diff
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|
2469 * files are affected |
|
279d0427ef26
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1
diff
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|
2470 */ |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
2471 if(data->decoded_all) |
|
279d0427ef26
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parents:
1
diff
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|
2472 { |
|
279d0427ef26
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parents:
1
diff
changeset
|
2473 /* Corner Case: Now that play calls are pushed |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
2474 * off to update(), the start call must |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
2475 * also come through here. So, start loops |
|
279d0427ef26
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parents:
1
diff
changeset
|
2476 * must be +1 |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
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|
2477 */ |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
2478 if(-1 == loops) |
|
279d0427ef26
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|
2479 { |
|
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|
2480 /* -1 is a special case, and you don't want |
|
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diff
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|
2481 * to add +1 to it */ |
|
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diff
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|
2482 ALmixer_Channel_List[channel].loops = -1; |
|
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|
2483 alSourcei(ALmixer_Channel_List[channel].alsource, AL_LOOPING, AL_TRUE); |
|
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|
2484 } |
|
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|
2485 else |
|
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|
2486 { |
|
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|
2487 ALmixer_Channel_List[channel].loops = loops+1; |
|
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diff
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|
2488 alSourcei(ALmixer_Channel_List[channel].alsource, AL_LOOPING, AL_FALSE); |
|
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2489 } |
|
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2490 } |
|
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|
2491 else |
|
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|
2492 { |
|
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|
2493 ALmixer_Channel_List[channel].loops = loops; |
|
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|
2494 /* Can we really loop on streamed data? */ |
|
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|
2495 alSourcei(ALmixer_Channel_List[channel].alsource, AL_LOOPING, AL_TRUE); |
|
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|
2496 } |
|
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|
2497 #endif |
|
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|
2498 |
|
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|
2499 /* Should I start playing here or pass the buck to update? */ |
|
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|
2500 /* Unlike SDL_SoundMixer, I think I'll do it here because |
|
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|
2501 * this library isn't a *total* hack and OpenAL has more |
|
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|
2502 * built in functionality I need, so less needs to be |
|
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|
2503 * controlled and directed through the update function. |
|
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|
2504 * The downside is less functionality is centralized. |
|
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|
2505 * The upside is that the update function should be |
|
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2506 * easier to maintain. |
|
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|
2507 */ |
|
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|
2508 |
|
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|
2509 /* Clear the error flag */ |
|
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|
2510 alGetError(); |
|
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|
2511 if(data->decoded_all) |
|
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|
2512 { |
|
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diff
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|
2513 /* Bind the data to the source */ |
|
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|
2514 alSourcei( |
|
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|
2515 ALmixer_Channel_List[channel].alsource, |
|
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|
2516 AL_BUFFER, |
|
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|
2517 data->buffer[0]); |
|
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diff
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|
2518 if((error = alGetError()) != AL_NO_ERROR) |
|
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|
2519 { |
|
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|
2520 ALmixer_SetError("Could not bind data to source: %s", |
|
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|
2521 alGetString(error) ); |
|
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|
2522 Clean_Channel(channel); |
|
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|
2523 return -1; |
|
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|
2524 } |
|
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|
2525 |
|
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diff
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|
2526 /* Make data available if access_data is enabled */ |
|
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diff
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|
2527 Invoke_Predecoded_Channel_Data_Callback(channel, data); |
|
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|
2528 } |
|
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|
2529 else |
|
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|
2530 { |
|
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|
2531 /* Need to use the streaming buffer for binding */ |
|
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diff
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|
2532 |
|
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|
2533 ALuint bytes_returned; |
|
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|
2534 ALuint j; |
|
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|
2535 data->num_buffers_in_use=0; |
|
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diff
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|
2536 /****** MODIFICATION must go here *********/ |
|
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|
2537 /* Since buffer queuing is pushed off until here to |
|
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diff
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|
2538 * avoid buffer conflicts, we must start reading |
|
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diff
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|
2539 * data here. First we make sure we have at least one |
|
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diff
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|
2540 * packet. Then we queue up until we hit our limit. |
|
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|
2541 */ |
|
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|
2542 bytes_returned = GetMoreData( |
|
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|
2543 data, |
|
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diff
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|
2544 data->buffer[0]); |
|
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|
2545 if(0 == bytes_returned) |
|
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|
2546 { |
|
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|
2547 /* No data or error */ |
|
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diff
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|
2548 ALmixer_SetError("Could not get data for streamed PlayChannel: %s", ALmixer_GetError()); |
|
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|
2549 Clean_Channel(channel); |
|
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|
2550 return -1; |
|
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|
2551 } |
|
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|
2552 /* Increment the number of buffers in use */ |
|
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|
2553 data->num_buffers_in_use++; |
|
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|
2554 |
|
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|
2555 |
|
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|
2556 /* Now we need to fill up the rest of the buffers. |
|
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|
2557 * There is a corner case where we run out of data |
|
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|
2558 * before the last buffer is filled. |
|
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|
2559 * Stop conditions are we run out of |
|
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|
2560 * data or we max out our preload buffers. |
|
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|
2561 */ |
|
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|
2562 |
|
6
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|
2563 /* |
|
2
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|
2564 fprintf(stderr, "Filling buffer #%d (AL id is %d)\n", 0, data->buffer[0]); |
|
6
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|
2565 */ |
|
2
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|
2566 for(j=1; j<data->num_startup_buffers; j++) |
|
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|
2567 { |
|
6
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|
2568 /* |
|
2
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|
2569 fprintf(stderr, "Filling buffer #%d (AL id is %d)\n", j, data->buffer[j]); |
|
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|
2570 fprintf(stderr, ">>>>>>>>>>>>>>>>>>HACK for GetMoreData2\n"); |
|
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|
2571 */ |
|
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|
2572 bytes_returned = GetMoreData( |
|
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|
2573 data, |
|
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|
2574 data->buffer[j]); |
|
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|
2575 /* |
|
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|
2576 * This might be a problem. I made a mistake with the types. I accidentally |
|
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|
2577 * made the bytes returned an ALint and returned -1 on error. |
|
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|
2578 * Bytes returned should be a ALuint, so now I no longer have a -1 case |
|
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|
2579 * to check. I hope I didn't break anything here |
|
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|
2580 */ |
|
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|
2581 #if 0 |
|
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|
2582 if(bytes_returned < 0) |
|
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|
2583 { |
|
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|
2584 /* Error found */ |
|
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|
2585 ALmixer_SetError("Could not get data for additional startup buffers for PlayChannel: %s", ALmixer_GetError()); |
|
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|
2586 /* We'll continue on because we do have some valid data */ |
|
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|
2587 ret_flag = -1; |
|
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|
2588 break; |
|
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|
2589 } |
|
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|
2590 else if(0 == bytes_returned) |
|
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|
2591 #endif |
|
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|
2592 if(0 == bytes_returned) |
|
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|
2593 { |
|
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|
2594 /* No more data to buffer */ |
|
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1
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|
2595 /* Check for loops */ |
|
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2596 if( ALmixer_Channel_List[channel].loops != 0 ) |
|
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|
2597 { |
|
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|
2598 /* |
|
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|
2599 fprintf(stderr, "Need to rewind. In RAMPUP, handling loop\n"); |
|
6
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|
2600 */ |
|
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|
2601 if(0 == Sound_Rewind(data->sample)) |
|
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|
2602 { |
|
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|
2603 fprintf(stderr, "error in rewind\n"); |
|
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|
2604 ALmixer_SetError( Sound_GetError() ); |
|
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|
2605 ALmixer_Channel_List[channel].loops = 0; |
|
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diff
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|
2606 ret_flag = -1; |
|
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diff
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|
2607 /* We'll continue on because we do have some valid data */ |
|
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|
2608 break; |
|
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|
2609 } |
|
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|
2610 /* Remember to reset the data->eof flag */ |
|
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diff
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|
2611 data->eof = 0; |
|
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|
2612 if(ALmixer_Channel_List[channel].loops > 0) |
|
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|
2613 { |
|
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|
2614 ALmixer_Channel_List[channel].loops--; |
|
6
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|
2615 /* |
|
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|
2616 fprintf(stderr, "Inside 000 >>>>>>>>>>Loops=%d\n", ALmixer_Channel_List[channel].loops); |
|
6
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2617 */ |
|
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|
2618 } |
|
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|
2619 /* Would like to redo the loop, but due to |
|
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diff
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|
2620 * Sound_Rewind() bugs, we would risk falling |
|
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diff
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|
2621 * into an infinite loop |
|
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|
2622 */ |
|
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|
2623 bytes_returned = GetMoreData( |
|
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|
2624 data, |
|
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diff
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|
2625 data->buffer[j]); |
|
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2626 if(bytes_returned <= 0) |
|
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|
2627 { |
|
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|
2628 ALmixer_SetError("Could not get data: %s", ALmixer_GetError()); |
|
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|
2629 /* We'll continue on because we do have some valid data */ |
|
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|
2630 ret_flag = -1; |
|
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|
2631 break; |
|
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|
2632 } |
|
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2633 } |
|
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|
2634 else |
|
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|
2635 { |
|
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|
2636 /* No loops to do so quit here */ |
|
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diff
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|
2637 break; |
|
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|
2638 } |
|
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|
2639 } |
|
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|
2640 /* Increment the number of buffers in use */ |
|
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diff
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|
2641 data->num_buffers_in_use++; |
|
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|
2642 } |
|
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|
2643 /* |
|
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|
2644 fprintf(stderr, "In PlayChannel, about to queue: source=%d, num_buffers_in_use=%d\n", |
|
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|
2645 ALmixer_Channel_List[channel].alsource, |
|
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|
2646 data->num_buffers_in_use); |
|
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|
2647 */ |
|
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|
2648 |
|
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|
2649 alSourceQueueBuffers( |
|
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|
2650 ALmixer_Channel_List[channel].alsource, |
|
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|
2651 data->num_buffers_in_use, |
|
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diff
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|
2652 data->buffer); |
|
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diff
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|
2653 if((error = alGetError()) != AL_NO_ERROR) |
|
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2654 { |
|
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|
2655 ALmixer_SetError("Could not bind data to source: %s", |
|
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|
2656 alGetString(error) ); |
|
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diff
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|
2657 Clean_Channel(channel); |
|
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|
2658 return -1; |
|
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|
2659 } |
|
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diff
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|
2660 /* This is part of the hideous Nvidia workaround. In order to figure out |
|
279d0427ef26
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|
2661 * which buffer to show during callbacks (for things like |
|
279d0427ef26
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1
diff
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|
2662 * o-scopes), I must keep a copy of the buffers that are queued in my own |
|
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diff
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|
2663 * data structure. This code will be called only if |
|
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diff
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|
2664 * "access_data" was set, indicated by whether the queue is NULL. |
|
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|
2665 */ |
|
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diff
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|
2666 if(data->circular_buffer_queue != NULL) |
|
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|
2667 { |
|
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|
2668 ALuint k; |
|
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diff
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|
2669 ALuint queue_ret_flag; |
|
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|
2670 for(k=0; k<data->num_buffers_in_use; k++) |
|
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|
2671 { |
|
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diff
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|
2672 // fprintf(stderr, "56c: CircularQueue_PushBack.\n"); |
|
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diff
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|
2673 queue_ret_flag = CircularQueueUnsignedInt_PushBack(data->circular_buffer_queue, data->buffer[k]); |
|
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diff
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|
2674 if(0 == queue_ret_flag) |
|
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diff
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|
2675 { |
|
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diff
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|
2676 fprintf(stderr, "Serious internal error: CircularQueue could not push into queue.\n"); |
|
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|
2677 ALmixer_SetError("Serious internal error: CircularQueue failed to push into queue"); |
|
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|
2678 } |
|
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diff
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|
2679 /* |
|
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diff
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|
2680 else |
|
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|
2681 { |
|
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|
2682 fprintf(stderr, "Queue in PlayTimed\n"); |
|
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|
2683 CircularQueueUnsignedInt_Print(data->circular_buffer_queue); |
|
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2684 } |
|
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|
2685 */ |
|
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2686 } |
|
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2687 } |
|
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|
2688 |
|
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|
2689 |
|
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|
2690 /****** END **********/ |
|
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2691 } |
|
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|
2692 /* We have finished loading the data (predecoded or queued) |
|
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2693 * so now we can play |
|
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|
2694 */ |
|
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|
2695 alSourcePlay(ALmixer_Channel_List[channel].alsource); |
|
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|
2696 if((error = alGetError()) != AL_NO_ERROR) |
|
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2697 { |
|
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|
2698 ALmixer_SetError("Play failed: %s", |
|
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|
2699 alGetString(error) ); |
|
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|
2700 Clean_Channel(channel); |
|
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2701 return -1; |
|
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|
2702 } |
|
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|
2703 |
|
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|
2704 /* Add to the counter that something is playing */ |
|
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|
2705 Is_Playing_global++; |
|
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2706 if(-1 == ret_flag) |
|
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|
2707 { |
|
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|
2708 fprintf(stderr, "BACKDOOR ERROR >>>>>>>>>>>>>>>>>>\n"); |
|
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|
2709 return -1; |
|
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|
2710 } |
|
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|
2711 return channel; |
|
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|
2712 } |
|
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2713 |
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2714 |
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2715 /* In case the user wants to specify a source instead of a channel, |
|
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2716 * they may use this function. This function will look up the |
|
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2717 * source-to-channel map, and convert the call into a |
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2718 * PlayChannelTimed() function call. |
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2719 * Returns the channel it's being played on. |
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2720 * Note: If you are prefer this method, then you need to be careful |
|
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2721 * about using PlayChannel, particularly if you request the |
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2722 * first available channels because source and channels have |
|
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2723 * a one-to-one mapping in this API. It is quite easy for |
|
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2724 * a channel/source to already be in use because of this. |
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2725 * In this event, an error message will be returned to you. |
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2726 */ |
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2727 static ALuint Internal_PlaySourceTimed(ALuint source, ALmixer_Data* data, ALint loops, ALint ticks) |
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2728 { |
|
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2729 ALint channel; |
|
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2730 ALint retval; |
|
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2731 if(0 == source) |
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2732 { |
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2733 retval = Internal_PlayChannelTimed(-1, data, loops, ticks); |
|
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2734 if(-1 == retval) |
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2735 { |
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2736 return 0; |
|
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2737 } |
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2738 else |
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2739 { |
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2740 return Internal_GetSource(retval); |
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2741 } |
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2742 } |
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2743 |
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2744 channel = Internal_GetChannel(source); |
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2745 if(-1 == channel) |
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2746 { |
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2747 ALmixer_SetError("Cannot Play source: %s", ALmixer_GetError()); |
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2748 return 0; |
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2749 } |
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2750 retval = Internal_PlayChannelTimed(channel, data, loops, ticks); |
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2751 if(-1 == retval) |
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2752 { |
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2753 return 0; |
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2754 } |
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2755 else |
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2756 { |
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2757 return source; |
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2758 } |
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2759 /* make compiler happy */ |
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2760 return 0; |
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2761 } |
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2762 |
| 0 | 2763 |
| 2764 | |
| 2765 | |
| 2766 /* Returns the channel or number of channels actually paused */ | |
| 2767 | |
|
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2768 static ALint Internal_PauseChannel(ALint channel) |
| 0 | 2769 { |
| 2770 ALenum error; | |
| 2771 ALint state; | |
|
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2772 ALint retval = 0; |
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2773 ALint counter = 0; |
| 0 | 2774 |
| 2775 if(channel >= Number_of_Channels_global) | |
| 2776 { | |
| 2777 ALmixer_SetError("Cannot pause channel %d because it exceeds maximum number of channels (%d)\n", channel, Number_of_Channels_global); | |
| 2778 return -1; | |
| 2779 } | |
| 2780 | |
| 2781 if((error = alGetError()) != AL_NO_ERROR) | |
| 2782 { | |
| 2783 fprintf(stderr, "28Testing error: %s\n", | |
|
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2784 alGetString(error)); |
| 0 | 2785 } |
| 2786 /* Clear error */ | |
| 2787 alGetError(); | |
| 2788 | |
| 2789 /* If the user specified a specific channel */ | |
| 2790 if(channel >= 0) | |
| 2791 { | |
| 2792 /* only need to process channel if in use */ | |
| 2793 if(ALmixer_Channel_List[channel].channel_in_use) | |
| 2794 { | |
| 2795 /* We don't want to repause if already | |
| 2796 * paused because the fadeout/expire | |
| 2797 * timing will get messed up | |
| 2798 */ | |
| 2799 alGetSourcei( | |
| 2800 ALmixer_Channel_List[channel].alsource, | |
| 2801 AL_SOURCE_STATE, &state | |
| 2802 ); | |
| 2803 if((error = alGetError()) != AL_NO_ERROR) | |
| 2804 { | |
| 2805 fprintf(stderr, "29Testing error: %s\n", | |
|
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2806 alGetString(error)); |
| 0 | 2807 } |
| 2808 if(AL_PLAYING == state) | |
| 2809 { | |
| 2810 /* Count the actual number of channels being paused */ | |
| 2811 counter++; | |
| 2812 | |
| 2813 alSourcePause(ALmixer_Channel_List[channel].alsource); | |
| 2814 if((error = alGetError()) != AL_NO_ERROR) | |
| 2815 { | |
| 2816 ALmixer_SetError("%s", | |
|
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2817 alGetString(error) ); |
| 0 | 2818 retval = -1; |
| 2819 } | |
| 2820 /* We need to pause the expire time count down */ | |
| 2821 if(ALmixer_Channel_List[channel].expire_ticks != -1) | |
| 2822 { | |
|
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2823 ALuint current_time = ALmixer_GetTicks(); |
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2824 ALuint diff_time; |
| 0 | 2825 diff_time = current_time - |
| 2826 ALmixer_Channel_List[channel].start_time; | |
| 2827 /* When we unpause, we will want to reset | |
| 2828 * the start time so we can continue | |
| 2829 * to base calculations off GetTicks(). | |
| 2830 * This means we need to subtract the amount | |
| 2831 * of time already used up from expire_ticks. | |
| 2832 */ | |
| 2833 ALmixer_Channel_List[channel].expire_ticks = | |
| 2834 ALmixer_Channel_List[channel].expire_ticks - | |
| 2835 diff_time; | |
| 2836 /* Because -1 is a special value, we can't | |
| 2837 * allow the time to go negative | |
| 2838 */ | |
| 2839 if(ALmixer_Channel_List[channel].expire_ticks < 0) | |
| 2840 { | |
| 2841 ALmixer_Channel_List[channel].expire_ticks = 0; | |
| 2842 } | |
| 2843 } | |
| 2844 /* Do the same as expire time for fading */ | |
| 2845 if(ALmixer_Channel_List[channel].fade_enabled) | |
| 2846 { | |
|
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2847 ALuint current_time = ALmixer_GetTicks(); |
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2848 ALuint diff_time; |
| 0 | 2849 diff_time = current_time - |
| 2850 ALmixer_Channel_List[channel].fade_start_time; | |
| 2851 /* When we unpause, we will want to reset | |
| 2852 * the start time so we can continue | |
| 2853 * to base calculations off GetTicks(). | |
| 2854 * This means we need to subtract the amount | |
| 2855 * of time already used up from expire_ticks. | |
| 2856 */ | |
| 2857 ALmixer_Channel_List[channel].fade_expire_ticks = | |
| 2858 ALmixer_Channel_List[channel].fade_expire_ticks - | |
| 2859 diff_time; | |
| 2860 /* Don't allow the time to go negative */ | |
| 2861 if(ALmixer_Channel_List[channel].expire_ticks < 0) | |
| 2862 { | |
| 2863 ALmixer_Channel_List[channel].expire_ticks = 0; | |
| 2864 } | |
| 2865 } /* End fade check */ | |
| 2866 } /* End if PLAYING */ | |
| 2867 } /* End If in use */ | |
| 2868 } /* End specific channel */ | |
| 2869 /* The user wants to halt all channels */ | |
| 2870 else | |
| 2871 { | |
|
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2872 ALint i; |
| 0 | 2873 for(i=0; i<Number_of_Channels_global; i++) |
| 2874 { | |
| 2875 /* only need to process channel if in use */ | |
| 2876 if(ALmixer_Channel_List[i].channel_in_use) | |
| 2877 { | |
| 2878 /* We don't want to repause if already | |
| 2879 * paused because the fadeout/expire | |
| 2880 * timing will get messed up | |
| 2881 */ | |
| 2882 alGetSourcei( | |
| 2883 ALmixer_Channel_List[i].alsource, | |
| 2884 AL_SOURCE_STATE, &state | |
| 2885 ); | |
| 2886 if((error = alGetError()) != AL_NO_ERROR) | |
| 2887 { | |
| 2888 fprintf(stderr, "30Testing error: %s\n", | |
|
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2889 alGetString(error)); |
| 0 | 2890 } |
| 2891 if(AL_PLAYING == state) | |
| 2892 { | |
| 2893 /* Count the actual number of channels being paused */ | |
| 2894 counter++; | |
| 2895 | |
| 2896 alSourcePause(ALmixer_Channel_List[i].alsource); | |
| 2897 if((error = alGetError()) != AL_NO_ERROR) | |
| 2898 { | |
| 2899 ALmixer_SetError("%s", | |
|
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2900 alGetString(error) ); |
| 0 | 2901 retval = -1; |
| 2902 } | |
| 2903 /* We need to pause the expire time count down */ | |
| 2904 if(ALmixer_Channel_List[i].expire_ticks != -1) | |
| 2905 { | |
|
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2906 ALuint current_time = ALmixer_GetTicks(); |
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2907 ALuint diff_time; |
| 0 | 2908 diff_time = current_time - |
| 2909 ALmixer_Channel_List[i].start_time; | |
| 2910 /* When we unpause, we will want to reset | |
| 2911 * the start time so we can continue | |
| 2912 * to base calculations off GetTicks(). | |
| 2913 * This means we need to subtract the amount | |
| 2914 * of time already used up from expire_ticks. | |
| 2915 */ | |
| 2916 ALmixer_Channel_List[i].expire_ticks = | |
| 2917 ALmixer_Channel_List[i].expire_ticks - | |
| 2918 diff_time; | |
| 2919 /* Because -1 is a special value, we can't | |
| 2920 * allow the time to go negative | |
| 2921 */ | |
| 2922 if(ALmixer_Channel_List[i].expire_ticks < 0) | |
| 2923 { | |
| 2924 ALmixer_Channel_List[i].expire_ticks = 0; | |
| 2925 } | |
| 2926 } | |
| 2927 /* Do the same as expire time for fading */ | |
| 2928 if(ALmixer_Channel_List[i].fade_enabled) | |
| 2929 { | |
|
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2930 ALuint current_time = ALmixer_GetTicks(); |
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2931 ALuint diff_time; |
| 0 | 2932 diff_time = current_time - |
| 2933 ALmixer_Channel_List[i].fade_start_time; | |
| 2934 /* When we unpause, we will want to reset | |
| 2935 * the start time so we can continue | |
| 2936 * to base calculations off GetTicks(). | |
| 2937 * This means we need to subtract the amount | |
| 2938 * of time already used up from expire_ticks. | |
| 2939 */ | |
| 2940 ALmixer_Channel_List[i].fade_expire_ticks = | |
| 2941 ALmixer_Channel_List[i].fade_expire_ticks - | |
| 2942 diff_time; | |
| 2943 /* Don't allow the time to go negative */ | |
| 2944 if(ALmixer_Channel_List[i].expire_ticks < 0) | |
| 2945 { | |
| 2946 ALmixer_Channel_List[i].expire_ticks = 0; | |
| 2947 } | |
| 2948 } /* End fade check */ | |
| 2949 } /* End if PLAYING */ | |
| 2950 } /* End channel in use */ | |
| 2951 } /* End for-loop */ | |
| 2952 } | |
| 2953 if(-1 == retval) | |
| 2954 { | |
| 2955 return -1; | |
| 2956 } | |
| 2957 return counter; | |
| 2958 } | |
| 2959 | |
| 2960 /* Returns the channel or number of channels actually paused */ | |
|
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2961 static ALint Internal_PauseSource(ALuint source) |
|
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2962 { |
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2963 ALint channel; |
| 0 | 2964 if(0 == source) |
| 2965 { | |
| 2966 return Internal_PauseChannel(-1); | |
| 2967 } | |
| 2968 | |
| 2969 channel = Internal_GetChannel(source); | |
| 2970 if(-1 == channel) | |
| 2971 { | |
| 2972 ALmixer_SetError("Cannot pause source: %s", ALmixer_GetError()); | |
| 2973 return -1; | |
| 2974 } | |
| 2975 return Internal_PauseChannel(channel); | |
| 2976 } | |
| 2977 | |
| 2978 | |
| 2979 | |
|
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2980 static ALint Internal_ResumeChannel(ALint channel) |
| 0 | 2981 { |
| 2982 ALint state; | |
| 2983 ALenum error; | |
|
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2984 ALint retval = 0; |
|
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2985 ALint counter = 0; |
| 0 | 2986 |
| 2987 if(channel >= Number_of_Channels_global) | |
| 2988 { | |
| 2989 ALmixer_SetError("Cannot pause channel %d because it exceeds maximum number of channels (%d)\n", channel, Number_of_Channels_global); | |
| 2990 return -1; | |
| 2991 } | |
| 2992 | |
| 2993 if((error = alGetError()) != AL_NO_ERROR) | |
| 2994 { | |
| 2995 fprintf(stderr, "31Testing error: %s\n", | |
|
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|
2996 alGetString(error)); |
| 0 | 2997 } |
| 2998 /* Clear error */ | |
| 2999 alGetError(); | |
| 3000 | |
| 3001 /* If the user specified a specific channel */ | |
| 3002 if(channel >= 0) | |
| 3003 { | |
| 3004 /* only need to process channel if in use */ | |
| 3005 if(ALmixer_Channel_List[channel].channel_in_use) | |
| 3006 { | |
| 3007 alGetSourcei( | |
| 3008 ALmixer_Channel_List[channel].alsource, | |
| 3009 AL_SOURCE_STATE, &state | |
| 3010 ); | |
| 3011 if((error = alGetError()) != AL_NO_ERROR) | |
| 3012 { | |
| 3013 fprintf(stderr, "32Testing error: %s\n", | |
|
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3014 alGetString(error)); |
| 0 | 3015 } |
| 3016 if(AL_PAUSED == state) | |
| 3017 { | |
| 3018 /* Count the actual number of channels resumed */ | |
| 3019 counter++; | |
| 3020 | |
| 3021 /* We need to resume the expire time count down */ | |
| 3022 if(ALmixer_Channel_List[channel].expire_ticks != -1) | |
| 3023 { | |
|
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|
3024 ALmixer_Channel_List[channel].start_time = ALmixer_GetTicks(); |
| 0 | 3025 } |
| 3026 /* Do the same as expire time for fading */ | |
| 3027 if(ALmixer_Channel_List[channel].fade_enabled) | |
| 3028 { | |
|
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|
3029 ALmixer_Channel_List[channel].fade_start_time = ALmixer_GetTicks(); |
| 0 | 3030 } |
| 3031 | |
| 3032 alSourcePlay(ALmixer_Channel_List[channel].alsource); | |
| 3033 if((error = alGetError()) != AL_NO_ERROR) | |
| 3034 { | |
| 3035 ALmixer_SetError("%s", | |
|
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|
3036 alGetString(error) ); |
| 0 | 3037 retval = -1; |
| 3038 } | |
| 3039 } | |
| 3040 } | |
| 3041 } | |
| 3042 /* The user wants to halt all channels */ | |
| 3043 else | |
| 3044 { | |
|
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|
3045 ALint i; |
| 0 | 3046 for(i=0; i<Number_of_Channels_global; i++) |
| 3047 { | |
| 3048 /* only need to process channel if in use */ | |
| 3049 if(ALmixer_Channel_List[i].channel_in_use) | |
| 3050 { | |
| 3051 alGetSourcei( | |
| 3052 ALmixer_Channel_List[i].alsource, | |
| 3053 AL_SOURCE_STATE, &state | |
| 3054 ); | |
| 3055 if((error = alGetError()) != AL_NO_ERROR) | |
| 3056 { | |
| 3057 fprintf(stderr, "33Testing error: %s\n", | |
|
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3058 alGetString(error)); |
| 0 | 3059 } |
| 3060 if(AL_PAUSED == state) | |
| 3061 { | |
| 3062 /* Count the actual number of channels resumed */ | |
| 3063 counter++; | |
| 3064 | |
| 3065 /* We need to resume the expire time count down */ | |
| 3066 if(ALmixer_Channel_List[i].expire_ticks != -1) | |
| 3067 { | |
|
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|
3068 ALmixer_Channel_List[i].start_time = ALmixer_GetTicks(); |
| 0 | 3069 } |
| 3070 /* Do the same as expire time for fading */ | |
| 3071 if(ALmixer_Channel_List[i].fade_enabled) | |
| 3072 { | |
|
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|
3073 ALmixer_Channel_List[i].fade_start_time = ALmixer_GetTicks(); |
| 0 | 3074 } |
| 3075 | |
| 3076 alSourcePlay(ALmixer_Channel_List[i].alsource); | |
| 3077 if((error = alGetError()) != AL_NO_ERROR) | |
| 3078 { | |
| 3079 ALmixer_SetError("%s", | |
|
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|
3080 alGetString(error) ); |
| 0 | 3081 retval = -1; |
| 3082 } | |
| 3083 } | |
| 3084 } | |
| 3085 } | |
| 3086 } | |
| 3087 if(-1 == retval) | |
| 3088 { | |
| 3089 return -1; | |
| 3090 } | |
| 3091 return counter; | |
| 3092 } | |
| 3093 | |
| 3094 | |
|
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|
3095 static ALint Internal_ResumeSource(ALuint source) |
|
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3096 { |
|
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|
3097 ALint channel; |
| 0 | 3098 if(0 == source) |
| 3099 { | |
| 3100 return Internal_ResumeChannel(-1); | |
| 3101 } | |
| 3102 | |
| 3103 channel = Internal_GetChannel(source); | |
| 3104 if(-1 == channel) | |
| 3105 { | |
| 3106 ALmixer_SetError("Cannot resume source: %s", ALmixer_GetError()); | |
| 3107 return -1; | |
| 3108 } | |
| 3109 return Internal_ResumeChannel(channel); | |
| 3110 } | |
| 3111 | |
| 3112 | |
| 3113 /* Might consider setting eof to 0 as a "feature" | |
| 3114 * This will allow seek to end to stay there because | |
| 3115 * Play automatically rewinds if at the end */ | |
|
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3116 static ALboolean Internal_SeekData(ALmixer_Data* data, ALuint msec) |
|
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3117 { |
|
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|
3118 ALint retval; |
| 0 | 3119 |
| 3120 if(NULL == data) | |
| 3121 { | |
| 3122 ALmixer_SetError("Cannot Seek because data is NULL"); | |
|
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3123 return AL_FALSE; |
| 0 | 3124 } |
| 3125 | |
| 3126 /* Seek for predecoded files involves moving the chunk pointer around */ | |
| 3127 if(data->decoded_all) | |
| 3128 { | |
|
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3129 ALuint byte_position; |
| 0 | 3130 |
| 3131 /* OpenAL doesn't seem to like it if I change the buffer | |
| 3132 * while playing (crashes), so I must require that Seek only | |
| 3133 * be done when the data is not in use. | |
| 3134 * Since data may be shared among multiple sources, | |
| 3135 * I can't shut them down myself, so I have to return an error. | |
| 3136 */ | |
| 3137 if(data->in_use) | |
| 3138 { | |
| 3139 ALmixer_SetError("Cannot seek on predecoded data while instances are playing"); | |
|
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3140 return AL_FALSE; |
| 0 | 3141 } |
| 3142 #if 0 | |
| 3143 #if defined(DISABLE_PREDECODED_SEEK) | |
| 3144 ALmixer_SetError("Seek support for predecoded samples was not compiled in"); | |
|
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3145 return AL_FALSE; |
| 0 | 3146 |
| 3147 #elif !defined(DISABLE_SEEK_MEMORY_OPTIMIZATION) | |
| 3148 /* By default, ALmixer frees the Sound_Sample for predecoded | |
| 3149 * samples because of the potential memory waste. | |
| 3150 * However, to seek a sample, we need to have a full | |
| 3151 * copy of the data around. So the strategy is to | |
| 3152 * recreate a hackish Sound_Sample to be used for seeking | |
| 3153 * purposes. If Sound_Sample is NULL, we will reallocate | |
| 3154 * memory for it and then procede as if everything | |
| 3155 * was normal. | |
| 3156 */ | |
| 3157 if(NULL == data->sample) | |
| 3158 { | |
| 3159 if( -1 == Reconstruct_Sound_Sample(data) ) | |
| 3160 { | |
|
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3161 return AL_FALSE; |
| 0 | 3162 } |
| 3163 } | |
| 3164 #endif | |
| 3165 #endif | |
| 3166 /* If access_data was set, then we still have the | |
| 3167 * Sound_Sample and we can move around in the data. | |
| 3168 * If it was not set, the data has been freed and we | |
| 3169 * cannot do anything because there is no way to | |
| 3170 * recover the data because OpenAL won't let us | |
| 3171 * get access to the buffers | |
| 3172 */ | |
| 3173 if(NULL == data->sample) | |
| 3174 { | |
| 3175 ALmixer_SetError("Cannot seek because access_data flag was set false when data was initialized"); | |
|
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3176 return AL_FALSE; |
| 0 | 3177 } |
| 3178 | |
| 3179 byte_position = Convert_Msec_To_Byte_Pos(&data->sample->desired, msec); | |
|
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|
3180 retval = Set_Predecoded_Seek_Position(data, byte_position); |
|
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3181 if(-1 == retval) |
|
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3182 { |
|
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3183 return AL_FALSE; |
|
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|
3184 } |
|
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|
3185 else |
|
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|
3186 { |
|
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|
3187 return AL_TRUE; |
|
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|
3188 } |
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|
3189 |
| 0 | 3190 } |
| 3191 else | |
| 3192 { | |
| 3193 /* Reset eof flag?? */ | |
| 3194 data->eof = 0; | |
| 3195 retval = Sound_Seek(data->sample, msec); | |
| 3196 if(0 == retval) | |
| 3197 { | |
| 3198 ALmixer_SetError(Sound_GetError()); | |
| 3199 | |
| 3200 fprintf(stderr, "Sound seek error: %s\n", ALmixer_GetError()); | |
| 3201 /* Try rewinding to clean up? */ | |
| 3202 /* | |
| 3203 Internal_RewindData(data); | |
| 3204 */ | |
|
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|
3205 return AL_FALSE; |
|
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|
3206 } |
|
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|
3207 return AL_TRUE; |
|
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|
3208 } |
|
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|
3209 |
|
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|
3210 return AL_TRUE; |
| 0 | 3211 } |
| 3212 | |
| 3213 | |
| 3214 | |
|
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|
3215 static ALint Internal_FadeInChannelTimed(ALint channel, ALmixer_Data* data, ALint loops, ALuint fade_ticks, ALint expire_ticks) |
| 0 | 3216 { |
| 3217 ALfloat value; | |
| 3218 ALenum error; | |
| 3219 ALfloat original_value; | |
|
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|
3220 ALuint current_time = ALmixer_GetTicks(); |
|
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|
3221 ALint retval; |
| 0 | 3222 |
| 3223 | |
| 3224 | |
| 3225 if(channel >= Number_of_Channels_global) | |
| 3226 { | |
| 3227 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); | |
| 3228 return -1; | |
| 3229 } | |
| 3230 /* Let's call PlayChannelTimed to do the job. | |
| 3231 * There are two catches: | |
| 3232 * First is that we must set the volumes before the play call(s). | |
| 3233 * Second is that we must initialize the channel values | |
| 3234 */ | |
| 3235 | |
| 3236 if(channel < 0) | |
| 3237 { | |
| 3238 /* This might cause a problem for threads/race conditions. | |
| 3239 * We need to set the volume on an unknown channel, | |
| 3240 * so we need to request a channel first. Remember | |
| 3241 * that requesting a channel doesn't lock and it | |
| 3242 * could be surrendered to somebody else before we claim it. | |
| 3243 */ | |
| 3244 channel = Internal_GetChannel(0); | |
| 3245 if(-1 == channel) | |
| 3246 { | |
| 3247 return -1; | |
| 3248 } | |
| 3249 } | |
| 3250 else if(ALmixer_Channel_List[channel].channel_in_use) | |
| 3251 { | |
| 3252 ALmixer_SetError("Channel %d is already in use", channel); | |
| 3253 return -1; | |
| 3254 } | |
| 3255 | |
| 3256 | |
| 3257 /* Get the original volume in case of a problem */ | |
| 3258 alGetSourcef(ALmixer_Channel_List[channel].alsource, | |
|
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|
3259 AL_GAIN, &original_value); |
| 0 | 3260 |
| 3261 if((error = alGetError()) != AL_NO_ERROR) | |
| 3262 { | |
| 3263 fprintf(stderr, "35Testing error: %s\n", | |
|
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|
3264 alGetString(error)); |
|
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|
3265 } |
|
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|
3266 ALmixer_Channel_List[channel].fade_end_volume = original_value; |
| 0 | 3267 |
| 3268 /* Get the Min volume */ | |
| 3269 alGetSourcef(ALmixer_Channel_List[channel].alsource, | |
| 3270 AL_MIN_GAIN, &value); | |
| 3271 if((error = alGetError()) != AL_NO_ERROR) | |
| 3272 { | |
| 3273 fprintf(stderr, "36Testing error: %s\n", | |
|
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|
3274 alGetString(error)); |
| 0 | 3275 } |
| 3276 ALmixer_Channel_List[channel].fade_start_volume = value; | |
| 3277 | |
| 3278 /* Set the actual volume */ | |
| 3279 alSourcef(ALmixer_Channel_List[channel].alsource, | |
|
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|
3280 AL_GAIN, value); |
| 0 | 3281 if((error = alGetError()) != AL_NO_ERROR) |
| 3282 { | |
| 3283 fprintf(stderr, "37Testing error: %s\n", | |
|
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|
3284 alGetString(error)); |
| 0 | 3285 } |
| 3286 | |
| 3287 | |
| 3288 /* Now call PlayChannelTimed */ | |
| 3289 retval = Internal_PlayChannelTimed(channel, data, loops, expire_ticks); | |
| 3290 if(-1 == retval) | |
| 3291 { | |
| 3292 /* Chance of failure is actually pretty high since | |
| 3293 * a channel might already be in use or streamed | |
| 3294 * data can be shared | |
| 3295 */ | |
| 3296 /* Restore the original value to avoid accidental | |
| 3297 * distruption of playback | |
| 3298 */ | |
| 3299 alSourcef(ALmixer_Channel_List[channel].alsource, | |
|
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3300 AL_GAIN, original_value); |
|
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|
3301 if((error = alGetError()) != AL_NO_ERROR) |
|
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|
3302 { |
|
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|
3303 fprintf(stderr, "38Testing error: %s\n", |
|
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|
3304 alGetString(error)); |
|
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|
3305 } |
| 0 | 3306 return retval; |
| 3307 } | |
| 3308 | |
| 3309 /* We can't accept 0 as a value because of div-by-zero. | |
| 3310 * If zero, just call PlayChannelTimed at normal | |
| 3311 * volume | |
| 3312 */ | |
| 3313 if(0 == fade_ticks) | |
| 3314 { | |
| 3315 alSourcef(ALmixer_Channel_List[channel].alsource, | |
|
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|
3316 AL_GAIN, |
| 0 | 3317 ALmixer_Channel_List[channel].fade_end_volume |
| 3318 ); | |
|
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|
3319 if((error = alGetError()) != AL_NO_ERROR) |
|
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|
3320 { |
|
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|
3321 fprintf(stderr, "39Testing error: %s\n", |
|
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|
3322 alGetString(error)); |
|
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|
3323 } |
| 0 | 3324 |
| 3325 return retval; | |
| 3326 } | |
| 3327 | |
| 3328 /* Enable fading effects via the flag */ | |
| 3329 ALmixer_Channel_List[channel].fade_enabled = 1; | |
| 3330 /* Set fade start time */ | |
| 3331 ALmixer_Channel_List[channel].fade_start_time | |
| 3332 = ALmixer_Channel_List[channel].start_time; | |
| 3333 /* Set the fade expire ticks */ | |
| 3334 ALmixer_Channel_List[channel].fade_expire_ticks = fade_ticks; | |
| 3335 | |
| 3336 /* Set 1/(endtime-starttime) or 1/deltaT */ | |
| 3337 ALmixer_Channel_List[channel].fade_inv_time = 1.0f / fade_ticks; | |
| 3338 | |
| 3339 return retval; | |
| 3340 | |
| 3341 } | |
| 3342 | |
| 3343 | |
|
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|
3344 static ALuint Internal_FadeInSourceTimed(ALuint source, ALmixer_Data* data, ALint loops, ALuint fade_ticks, ALint expire_ticks) |
|
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|
3345 { |
|
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|
3346 ALint channel; |
|
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|
3347 ALint retval; |
| 0 | 3348 if(0 == source) |
| 3349 { | |
| 3350 retval = Internal_FadeInChannelTimed(-1, data, loops, fade_ticks, expire_ticks); | |
| 3351 if(-1 == retval) | |
| 3352 { | |
| 3353 return 0; | |
| 3354 } | |
| 3355 else | |
| 3356 { | |
| 3357 return Internal_GetSource(retval); | |
| 3358 } | |
| 3359 } | |
| 3360 | |
| 3361 channel = Internal_GetChannel(source); | |
| 3362 if(-1 == channel) | |
| 3363 { | |
| 3364 ALmixer_SetError("Cannot FadeIn source: %s", ALmixer_GetError()); | |
| 3365 return 0; | |
| 3366 } | |
| 3367 retval = Internal_FadeInChannelTimed(channel, data, loops, fade_ticks, expire_ticks); | |
| 3368 if(-1 == retval) | |
| 3369 { | |
| 3370 return 0; | |
| 3371 } | |
| 3372 else | |
| 3373 { | |
| 3374 return source; | |
| 3375 } | |
| 3376 /* make compiler happy */ | |
| 3377 return 0; | |
| 3378 } | |
| 3379 | |
| 3380 | |
| 3381 | |
| 3382 | |
| 3383 /* Will fade out currently playing channels. | |
| 3384 * It starts at the current volume level and goes down */ | |
|
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|
3385 static ALint Internal_FadeOutChannel(ALint channel, ALuint ticks) |
| 0 | 3386 { |
| 3387 ALfloat value; | |
| 3388 ALenum error; | |
|
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|
3389 ALuint current_time = ALmixer_GetTicks(); |
|
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|
3390 ALuint counter = 0; |
| 0 | 3391 |
| 3392 /* We can't accept 0 as a value because of div-by-zero. | |
| 3393 * If zero, just call Halt at normal | |
| 3394 * volume | |
| 3395 */ | |
| 3396 if(0 == ticks) | |
| 3397 { | |
|
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|
3398 return Internal_HaltChannel(channel, AL_FALSE); |
| 0 | 3399 } |
| 3400 | |
| 3401 | |
| 3402 if(channel >= Number_of_Channels_global) | |
| 3403 { | |
| 3404 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); | |
| 3405 return -1; | |
| 3406 } | |
| 3407 | |
| 3408 if(channel >= 0) | |
| 3409 { | |
| 3410 if(ALmixer_Channel_List[channel].channel_in_use) | |
| 3411 { | |
| 3412 /* Get the current volume */ | |
| 3413 alGetSourcef(ALmixer_Channel_List[channel].alsource, | |
|
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|
3414 AL_GAIN, &value); |
| 0 | 3415 ALmixer_Channel_List[channel].fade_start_volume = value; |
| 3416 if((error = alGetError()) != AL_NO_ERROR) | |
| 3417 { | |
| 3418 fprintf(stderr, "40Testing error: %s\n", | |
|
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|
3419 alGetString(error)); |
| 0 | 3420 } |
| 3421 | |
| 3422 /* Get the Min volume */ | |
| 3423 alGetSourcef(ALmixer_Channel_List[channel].alsource, | |
| 3424 AL_MIN_GAIN, &value); | |
| 3425 if((error = alGetError()) != AL_NO_ERROR) | |
| 3426 { | |
| 3427 fprintf(stderr, "41Testing error: %s\n", | |
|
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|
3428 alGetString(error)); |
| 0 | 3429 } |
| 3430 ALmixer_Channel_List[channel].fade_end_volume = value; | |
| 3431 | |
| 3432 /* Set expire start time */ | |
| 3433 ALmixer_Channel_List[channel].start_time = current_time; | |
| 3434 /* Set the expire ticks */ | |
| 3435 ALmixer_Channel_List[channel].expire_ticks = ticks; | |
| 3436 /* Set fade start time */ | |
| 3437 ALmixer_Channel_List[channel].fade_start_time = current_time; | |
| 3438 /* Set the fade expire ticks */ | |
| 3439 ALmixer_Channel_List[channel].fade_expire_ticks = ticks; | |
| 3440 /* Enable fading effects via the flag */ | |
| 3441 ALmixer_Channel_List[channel].fade_enabled = 1; | |
| 3442 | |
| 3443 /* Set 1/(endtime-starttime) or 1/deltaT */ | |
| 3444 ALmixer_Channel_List[channel].fade_inv_time = 1.0f / ticks; | |
| 3445 | |
| 3446 counter++; | |
| 3447 } | |
| 3448 } | |
| 3449 /* Else need to fade out all channels */ | |
| 3450 else | |
| 3451 { | |
|
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|
3452 ALint i; |
| 0 | 3453 for(i=0; i<Number_of_Channels_global; i++) |
| 3454 { | |
| 3455 if(ALmixer_Channel_List[i].channel_in_use) | |
| 3456 { | |
| 3457 /* Get the current volume */ | |
| 3458 alGetSourcef(ALmixer_Channel_List[i].alsource, | |
|
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|
3459 AL_GAIN, &value); |
| 0 | 3460 ALmixer_Channel_List[i].fade_start_volume = value; |
| 3461 if((error = alGetError()) != AL_NO_ERROR) | |
| 3462 { | |
| 3463 fprintf(stderr, "42Testing error: %s\n", | |
|
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|
3464 alGetString(error)); |
| 0 | 3465 } |
| 3466 | |
| 3467 /* Get the Min volume */ | |
| 3468 alGetSourcef(ALmixer_Channel_List[i].alsource, | |
| 3469 AL_MIN_GAIN, &value); | |
| 3470 if((error = alGetError()) != AL_NO_ERROR) | |
| 3471 { | |
| 3472 fprintf(stderr, "43Testing error: %s\n", | |
|
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|
3473 alGetString(error)); |
| 0 | 3474 } |
| 3475 ALmixer_Channel_List[i].fade_end_volume = value; | |
| 3476 | |
| 3477 /* Set expire start time */ | |
| 3478 ALmixer_Channel_List[i].start_time = current_time; | |
| 3479 /* Set the expire ticks */ | |
| 3480 ALmixer_Channel_List[i].expire_ticks = ticks; | |
| 3481 /* Set fade start time */ | |
| 3482 ALmixer_Channel_List[i].fade_start_time = current_time; | |
| 3483 /* Set the fade expire ticks */ | |
| 3484 ALmixer_Channel_List[i].fade_expire_ticks = ticks; | |
| 3485 /* Enable fading effects via the flag */ | |
| 3486 ALmixer_Channel_List[i].fade_enabled = 1; | |
| 3487 | |
| 3488 /* Set 1/(endtime-starttime) or 1/deltaT */ | |
| 3489 ALmixer_Channel_List[i].fade_inv_time = 1.0f / ticks; | |
| 3490 | |
| 3491 counter++; | |
| 3492 } | |
| 3493 } /* End for loop */ | |
| 3494 } | |
| 3495 return counter; | |
| 3496 } | |
| 3497 | |
| 3498 | |
|
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|
3499 static ALint Internal_FadeOutSource(ALuint source, ALuint ticks) |
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|
3500 { |
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|
3501 ALint channel; |
| 0 | 3502 if(0 == source) |
| 3503 { | |
| 3504 return Internal_FadeOutChannel(-1, ticks); | |
| 3505 } | |
| 3506 | |
| 3507 channel = Internal_GetChannel(source); | |
| 3508 if(-1 == channel) | |
| 3509 { | |
| 3510 ALmixer_SetError("Cannot FadeOut source: %s", ALmixer_GetError()); | |
| 3511 return -1; | |
| 3512 } | |
| 3513 return Internal_FadeOutChannel(channel, ticks); | |
| 3514 } | |
| 3515 | |
| 3516 | |
| 3517 /* Will fade currently playing channels. | |
| 3518 * It starts at the current volume level and go to target | |
| 3519 * Only affects channels that are playing | |
| 3520 */ | |
|
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|
3521 static ALint Internal_FadeChannel(ALint channel, ALuint ticks, ALfloat volume) |
| 0 | 3522 { |
| 3523 ALfloat value; | |
| 3524 ALenum error; | |
|
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|
3525 ALuint current_time = ALmixer_GetTicks(); |
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|
3526 ALuint counter = 0; |
| 0 | 3527 |
| 3528 if(channel >= Number_of_Channels_global) | |
| 3529 { | |
| 3530 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); | |
| 3531 return -1; | |
| 3532 } | |
| 3533 | |
| 3534 if(channel >= 0) | |
| 3535 { | |
| 3536 if(volume < ALmixer_Channel_List[channel].min_volume) | |
| 3537 { | |
| 3538 volume = ALmixer_Channel_List[channel].min_volume; | |
| 3539 } | |
| 3540 else if(volume > ALmixer_Channel_List[channel].max_volume) | |
| 3541 { | |
| 3542 volume = ALmixer_Channel_List[channel].max_volume; | |
| 3543 } | |
| 3544 | |
| 3545 if(ALmixer_Channel_List[channel].channel_in_use) | |
| 3546 { | |
| 3547 if(ticks > 0) | |
| 3548 { | |
| 3549 /* Get the current volume */ | |
| 3550 alGetSourcef(ALmixer_Channel_List[channel].alsource, | |
|
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|
3551 AL_GAIN, &value); |
| 0 | 3552 if((error = alGetError()) != AL_NO_ERROR) |
| 3553 { | |
| 3554 fprintf(stderr, "44Testing error: %s\n", | |
|
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|
3555 alGetString(error)); |
| 0 | 3556 } |
| 3557 ALmixer_Channel_List[channel].fade_start_volume = value; | |
| 3558 | |
| 3559 /* Set the target volume */ | |
| 3560 ALmixer_Channel_List[channel].fade_end_volume = volume; | |
| 3561 | |
| 3562 /* Set fade start time */ | |
| 3563 ALmixer_Channel_List[channel].fade_start_time = current_time; | |
| 3564 /* Set the fade expire ticks */ | |
| 3565 ALmixer_Channel_List[channel].fade_expire_ticks = ticks; | |
| 3566 /* Enable fading effects via the flag */ | |
| 3567 ALmixer_Channel_List[channel].fade_enabled = 1; | |
| 3568 | |
| 3569 /* Set 1/(endtime-starttime) or 1/deltaT */ | |
| 3570 ALmixer_Channel_List[channel].fade_inv_time = 1.0f / ticks; | |
| 3571 } | |
| 3572 else | |
| 3573 { | |
| 3574 alSourcef(ALmixer_Channel_List[channel].alsource, | |
|
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|
3575 AL_GAIN, volume); |
| 0 | 3576 if((error = alGetError()) != AL_NO_ERROR) |
| 3577 { | |
| 3578 fprintf(stderr, "45Testing error: %s\n", | |
|
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|
3579 alGetString(error)); |
| 0 | 3580 } |
| 3581 } | |
| 3582 counter++; | |
| 3583 } | |
| 3584 } | |
| 3585 /* Else need to fade out all channels */ | |
| 3586 else | |
| 3587 { | |
|
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|
3588 ALint i; |
| 0 | 3589 for(i=0; i<Number_of_Channels_global; i++) |
| 3590 { | |
| 3591 if(volume < ALmixer_Channel_List[i].min_volume) | |
| 3592 { | |
| 3593 volume = ALmixer_Channel_List[i].min_volume; | |
| 3594 } | |
| 3595 else if(volume > ALmixer_Channel_List[i].max_volume) | |
| 3596 { | |
| 3597 volume = ALmixer_Channel_List[i].max_volume; | |
| 3598 } | |
| 3599 | |
| 3600 if(ALmixer_Channel_List[i].channel_in_use) | |
| 3601 { | |
| 3602 if(ticks > 0) | |
| 3603 { | |
| 3604 /* Get the current volume */ | |
| 3605 alGetSourcef(ALmixer_Channel_List[i].alsource, | |
|
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|
3606 AL_GAIN, &value); |
| 0 | 3607 if((error = alGetError()) != AL_NO_ERROR) |
| 3608 { | |
| 3609 fprintf(stderr, "46Testing error: %s\n", | |
|
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|
3610 alGetString(error)); |
| 0 | 3611 } |
| 3612 ALmixer_Channel_List[i].fade_start_volume = value; | |
| 3613 | |
| 3614 /* Set target volume */ | |
| 3615 ALmixer_Channel_List[i].fade_end_volume = volume; | |
| 3616 | |
| 3617 /* Set fade start time */ | |
| 3618 ALmixer_Channel_List[i].fade_start_time = current_time; | |
| 3619 /* Set the fade expire ticks */ | |
| 3620 ALmixer_Channel_List[i].fade_expire_ticks = ticks; | |
| 3621 /* Enable fading effects via the flag */ | |
| 3622 ALmixer_Channel_List[i].fade_enabled = 1; | |
| 3623 | |
| 3624 /* Set 1/(endtime-starttime) or 1/deltaT */ | |
| 3625 ALmixer_Channel_List[i].fade_inv_time = 1.0f / ticks; | |
| 3626 } | |
| 3627 else | |
| 3628 { | |
| 3629 alSourcef(ALmixer_Channel_List[i].alsource, | |
|
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|
3630 AL_GAIN, volume); |
| 0 | 3631 if((error = alGetError()) != AL_NO_ERROR) |
| 3632 { | |
| 3633 fprintf(stderr, "47Testing error: %s\n", | |
|
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|
3634 alGetString(error)); |
| 0 | 3635 } |
| 3636 } | |
| 3637 counter++; | |
| 3638 } | |
| 3639 } /* End for loop */ | |
| 3640 } | |
| 3641 return counter; | |
| 3642 } | |
| 3643 | |
|
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|
3644 static ALint Internal_FadeSource(ALuint source, ALuint ticks, ALfloat volume) |
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|
3645 { |
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|
3646 ALint channel; |
| 0 | 3647 if(0 == source) |
| 3648 { | |
| 3649 return Internal_FadeChannel(-1, ticks, volume); | |
| 3650 } | |
| 3651 | |
| 3652 channel = Internal_GetChannel(source); | |
| 3653 if(-1 == channel) | |
| 3654 { | |
| 3655 ALmixer_SetError("Cannot Fade source: %s", ALmixer_GetError()); | |
| 3656 return -1; | |
| 3657 } | |
| 3658 return Internal_FadeChannel(channel, ticks, volume); | |
| 3659 } | |
| 3660 | |
| 3661 | |
| 3662 | |
| 3663 | |
| 3664 /* Set a volume regardless if it's in use or not. | |
| 3665 */ | |
|
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|
3666 static ALboolean Internal_SetVolumeChannel(ALint channel, ALfloat volume) |
| 0 | 3667 { |
| 3668 ALenum error; | |
|
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|
3669 ALboolean retval = AL_TRUE; |
|
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|
3670 |
|
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|
3671 if(channel >= Number_of_Channels_global) |
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|
3672 { |
|
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|
3673 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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|
3674 return AL_FALSE; |
|
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|
3675 } |
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|
3676 |
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|
3677 if(channel >= 0) |
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|
3678 { |
|
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|
3679 if(volume < 0.0f) |
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|
3680 { |
|
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|
3681 volume = 0.0f; |
|
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|
3682 } |
|
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|
3683 else if(volume > 1.0f) |
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|
3684 { |
|
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|
3685 volume = 1.0f; |
|
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|
3686 } |
|
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|
3687 alSourcef(ALmixer_Channel_List[channel].alsource, |
|
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|
3688 AL_GAIN, volume); |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3689 if((error = alGetError()) != AL_NO_ERROR) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3690 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3691 ALmixer_SetError("%s", |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3692 alGetString(error) ); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3693 retval = AL_FALSE; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3694 } |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3695 } |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3696 else |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3697 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3698 ALint i; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3699 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
|
3700 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3701 if(volume < 0.0f) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3702 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3703 volume = 0.0f; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3704 } |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3705 else if(volume > 1.0f) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3706 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3707 volume = 1.0f; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3708 } |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3709 alSourcef(ALmixer_Channel_List[i].alsource, |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3710 AL_GAIN, volume); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3711 if((error = alGetError()) != AL_NO_ERROR) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3712 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3713 ALmixer_SetError("%s", |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3714 alGetString(error) ); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3715 retval = AL_FALSE; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3716 } |
|
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 } |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3719 return retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3720 } |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3721 |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3722 static ALboolean Internal_SetVolumeSource(ALuint source, ALfloat volume) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3723 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3724 ALint channel; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3725 if(0 == source) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3726 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3727 return Internal_SetVolumeChannel(-1, volume); |
|
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 |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3730 channel = Internal_GetChannel(source); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3731 if(-1 == channel) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3732 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3733 ALmixer_SetError("Cannot SetMaxVolume: %s", ALmixer_GetError()); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3734 return AL_FALSE; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3735 } |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3736 return Internal_SetVolumeChannel(channel, volume); |
|
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 |
|
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 static ALfloat Internal_GetVolumeChannel(ALint channel) |
|
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 ALfloat value; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3743 ALenum error; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3744 ALfloat running_total = 0.0f; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3745 ALfloat retval = 0.0f; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3746 |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3747 if(channel >= Number_of_Channels_global) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3748 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3749 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
|
3750 return -1.0f; |
|
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 |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3753 if(channel >= 0) |
|
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 alGetSourcef(ALmixer_Channel_List[channel].alsource, |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3756 AL_GAIN, &value); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3757 if((error = alGetError()) != AL_NO_ERROR) |
|
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 ALmixer_SetError("%s", alGetString(error) ); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3760 retval = -1.0f; |
|
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 else |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3763 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3764 retval = value; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3765 } |
|
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 else |
|
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 ALint i; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3770 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
|
3771 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3772 alGetSourcef(ALmixer_Channel_List[i].alsource, |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3773 AL_GAIN, &value); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3774 if((error = alGetError()) != AL_NO_ERROR) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3775 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3776 ALmixer_SetError("%s", alGetString(error) ); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3777 retval = -1; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3778 } |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3779 else |
|
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 running_total += value; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3782 } |
|
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 if(0 == Number_of_Channels_global) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3785 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3786 ALmixer_SetError("No channels are allocated"); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3787 retval = -1.0f; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3788 } |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3789 else |
|
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 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
|
3792 } |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3793 } |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3794 return retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3795 } |
|
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 static ALfloat Internal_GetVolumeSource(ALuint source) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3798 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3799 ALint channel; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3800 if(0 == source) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3801 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3802 return Internal_GetVolumeChannel(-1); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3803 } |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3804 |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3805 channel = Internal_GetChannel(source); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3806 if(-1 == channel) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3807 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3808 ALmixer_SetError("Cannot GetVolume: %s", ALmixer_GetError()); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
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3809 return -1.0f; |
|
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3810 } |
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3811 |
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3812 return Internal_GetVolumeChannel(channel); |
|
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3813 } |
|
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3814 |
|
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3815 |
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3816 |
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|
3817 /* Set a volume regardless if it's in use or not. |
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|
3818 */ |
|
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|
3819 static ALboolean Internal_SetMaxVolumeChannel(ALint channel, ALfloat volume) |
|
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|
3820 { |
|
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|
3821 ALenum error; |
|
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|
3822 ALboolean retval = AL_TRUE; |
| 0 | 3823 |
| 3824 if(channel >= Number_of_Channels_global) | |
| 3825 { | |
| 3826 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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|
3827 return AL_FALSE; |
| 0 | 3828 } |
| 3829 | |
| 3830 if(channel >= 0) | |
| 3831 { | |
| 3832 if(volume < 0.0f) | |
| 3833 { | |
| 3834 volume = 0.0f; | |
| 3835 } | |
| 3836 else if(volume > 1.0f) | |
| 3837 { | |
| 3838 volume = 1.0f; | |
| 3839 } | |
| 3840 ALmixer_Channel_List[channel].max_volume = volume; | |
| 3841 alSourcef(ALmixer_Channel_List[channel].alsource, | |
| 3842 AL_MAX_GAIN, volume); | |
| 3843 if((error = alGetError()) != AL_NO_ERROR) | |
| 3844 { | |
| 3845 ALmixer_SetError("%s", | |
|
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3846 alGetString(error) ); |
|
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|
3847 retval = AL_FALSE; |
| 0 | 3848 } |
| 3849 if(ALmixer_Channel_List[channel].max_volume < ALmixer_Channel_List[channel].min_volume) | |
| 3850 { | |
| 3851 ALmixer_Channel_List[channel].min_volume = volume; | |
| 3852 alSourcef(ALmixer_Channel_List[channel].alsource, | |
| 3853 AL_MIN_GAIN, volume); | |
| 3854 if((error = alGetError()) != AL_NO_ERROR) | |
| 3855 { | |
| 3856 ALmixer_SetError("%s", | |
|
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3857 alGetString(error) ); |
|
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|
3858 retval = AL_FALSE; |
| 0 | 3859 } |
| 3860 } | |
| 3861 } | |
| 3862 else | |
| 3863 { | |
|
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3864 ALint i; |
| 0 | 3865 for(i=0; i<Number_of_Channels_global; i++) |
| 3866 { | |
| 3867 if(volume < 0.0f) | |
| 3868 { | |
| 3869 volume = 0.0f; | |
| 3870 } | |
| 3871 else if(volume > 1.0f) | |
| 3872 { | |
| 3873 volume = 1.0f; | |
| 3874 } | |
| 3875 ALmixer_Channel_List[i].max_volume = volume; | |
| 3876 alSourcef(ALmixer_Channel_List[i].alsource, | |
| 3877 AL_MAX_GAIN, volume); | |
| 3878 if((error = alGetError()) != AL_NO_ERROR) | |
| 3879 { | |
| 3880 ALmixer_SetError("%s", | |
|
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|
3881 alGetString(error) ); |
|
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|
3882 retval = AL_FALSE; |
| 0 | 3883 } |
| 3884 if(ALmixer_Channel_List[i].max_volume < ALmixer_Channel_List[i].min_volume) | |
| 3885 { | |
| 3886 ALmixer_Channel_List[i].min_volume = volume; | |
| 3887 alSourcef(ALmixer_Channel_List[i].alsource, | |
| 3888 AL_MIN_GAIN, volume); | |
| 3889 if((error = alGetError()) != AL_NO_ERROR) | |
| 3890 { | |
| 3891 ALmixer_SetError("%s", | |
|
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|
3892 alGetString(error) ); |
|
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|
3893 retval = AL_FALSE; |
| 0 | 3894 } |
| 3895 } | |
| 3896 } | |
| 3897 } | |
| 3898 return retval; | |
| 3899 } | |
| 3900 | |
|
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3901 static ALint Internal_SetMaxVolumeSource(ALuint source, ALfloat volume) |
|
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3902 { |
|
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|
3903 ALint channel; |
| 0 | 3904 if(0 == source) |
| 3905 { | |
| 3906 return Internal_SetMaxVolumeChannel(-1, volume); | |
| 3907 } | |
| 3908 | |
| 3909 channel = Internal_GetChannel(source); | |
| 3910 if(-1 == channel) | |
| 3911 { | |
| 3912 ALmixer_SetError("Cannot SetMaxVolume: %s", ALmixer_GetError()); | |
|
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3913 return AL_FALSE; |
| 0 | 3914 } |
| 3915 return Internal_SetMaxVolumeChannel(channel, volume); | |
| 3916 } | |
| 3917 | |
|
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3918 static ALfloat Internal_GetMaxVolumeChannel(ALint channel) |
| 0 | 3919 { |
| 3920 /* | |
| 3921 ALfloat value; | |
| 3922 ALenum error; | |
| 3923 */ | |
| 3924 ALfloat running_total = 0.0f; | |
| 3925 ALfloat retval = 0.0f; | |
| 3926 | |
| 3927 if(channel >= Number_of_Channels_global) | |
| 3928 { | |
| 3929 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); | |
| 3930 return -1.0f; | |
| 3931 } | |
| 3932 | |
| 3933 if(channel >= 0) | |
| 3934 { | |
| 3935 /* | |
| 3936 alGetSourcef(ALmixer_Channel_List[channel].alsource, | |
| 3937 AL_GAIN, &value); | |
| 3938 if((error = alGetError()) != AL_NO_ERROR) | |
| 3939 { | |
| 3940 ALmixer_SetError("%s", | |
|
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3941 alGetString(error) ); |
| 0 | 3942 retval = -1.0f; |
| 3943 } | |
| 3944 else | |
| 3945 { | |
| 3946 retval = value; | |
| 3947 } | |
| 3948 */ | |
| 3949 retval = ALmixer_Channel_List[channel].max_volume; | |
| 3950 | |
| 3951 } | |
| 3952 else | |
| 3953 { | |
|
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3954 ALint i; |
| 0 | 3955 for(i=0; i<Number_of_Channels_global; i++) |
| 3956 { | |
| 3957 /* | |
| 3958 alGetSourcef(ALmixer_Channel_List[i].alsource, | |
| 3959 AL_GAIN, &value); | |
| 3960 if((error = alGetError()) != AL_NO_ERROR) | |
| 3961 { | |
| 3962 ALmixer_SetError("%s", | |
|
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3963 alGetString(error) ); |
| 0 | 3964 retval = -1; |
| 3965 } | |
| 3966 else | |
| 3967 { | |
| 3968 running_total += value; | |
| 3969 } | |
| 3970 */ | |
| 3971 running_total += ALmixer_Channel_List[i].max_volume; | |
| 3972 } | |
| 3973 if(0 == Number_of_Channels_global) | |
| 3974 { | |
| 3975 ALmixer_SetError("No channels are allocated"); | |
| 3976 retval = -1.0f; | |
| 3977 } | |
| 3978 else | |
| 3979 { | |
| 3980 retval = running_total / Number_of_Channels_global; | |
| 3981 } | |
| 3982 } | |
| 3983 return retval; | |
| 3984 } | |
| 3985 | |
| 3986 static ALfloat Internal_GetMaxVolumeSource(ALuint source) | |
| 3987 { | |
|
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3988 ALint channel; |
| 0 | 3989 if(0 == source) |
| 3990 { | |
| 3991 return Internal_GetMaxVolumeChannel(-1); | |
| 3992 } | |
| 3993 | |
| 3994 channel = Internal_GetChannel(source); | |
| 3995 if(-1 == channel) | |
| 3996 { | |
| 3997 ALmixer_SetError("Cannot GetVolume: %s", ALmixer_GetError()); | |
| 3998 return -1.0f; | |
| 3999 } | |
| 4000 | |
| 4001 return Internal_GetMaxVolumeChannel(channel); | |
| 4002 } | |
| 4003 | |
| 4004 | |
| 4005 /* Set a volume regardless if it's in use or not. | |
| 4006 */ | |
|
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4007 static ALboolean Internal_SetMinVolumeChannel(ALint channel, ALfloat volume) |
| 0 | 4008 { |
| 4009 ALenum error; | |
|
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4010 ALboolean retval = AL_TRUE; |
| 0 | 4011 |
| 4012 if(channel >= Number_of_Channels_global) | |
| 4013 { | |
| 4014 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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4015 return AL_FALSE; |
| 0 | 4016 } |
| 4017 | |
| 4018 if(channel >= 0) | |
| 4019 { | |
| 4020 if(volume < 0.0f) | |
| 4021 { | |
| 4022 volume = 0.0f; | |
| 4023 } | |
| 4024 else if(volume > 1.0f) | |
| 4025 { | |
| 4026 volume = 1.0f; | |
| 4027 } | |
| 4028 ALmixer_Channel_List[channel].min_volume = volume; | |
| 4029 alSourcef(ALmixer_Channel_List[channel].alsource, | |
| 4030 AL_MIN_GAIN, volume); | |
| 4031 if((error = alGetError()) != AL_NO_ERROR) | |
| 4032 { | |
| 4033 ALmixer_SetError("%s", | |
|
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|
4034 alGetString(error) ); |
|
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|
4035 retval = AL_FALSE; |
| 0 | 4036 } |
| 4037 if(ALmixer_Channel_List[channel].max_volume < ALmixer_Channel_List[channel].min_volume) | |
| 4038 { | |
| 4039 ALmixer_Channel_List[channel].max_volume = volume; | |
| 4040 alSourcef(ALmixer_Channel_List[channel].alsource, | |
| 4041 AL_MAX_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 } | |
| 4049 } | |
| 4050 else | |
| 4051 { | |
|
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4052 ALint i; |
| 0 | 4053 for(i=0; i<Number_of_Channels_global; i++) |
| 4054 { | |
| 4055 if(volume < 0.0f) | |
| 4056 { | |
| 4057 volume = 0.0f; | |
| 4058 } | |
| 4059 else if(volume > 1.0f) | |
| 4060 { | |
| 4061 volume = 1.0f; | |
| 4062 } | |
| 4063 ALmixer_Channel_List[i].min_volume = volume; | |
| 4064 alSourcef(ALmixer_Channel_List[i].alsource, | |
| 4065 AL_MIN_GAIN, volume); | |
| 4066 if((error = alGetError()) != AL_NO_ERROR) | |
| 4067 { | |
| 4068 ALmixer_SetError("%s", | |
|
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|
4069 alGetString(error) ); |
|
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|
4070 retval = AL_FALSE; |
| 0 | 4071 } |
| 4072 if(ALmixer_Channel_List[i].max_volume < ALmixer_Channel_List[i].min_volume) | |
| 4073 { | |
| 4074 ALmixer_Channel_List[i].max_volume = volume; | |
| 4075 alSourcef(ALmixer_Channel_List[i].alsource, | |
| 4076 AL_MAX_GAIN, volume); | |
| 4077 if((error = alGetError()) != AL_NO_ERROR) | |
| 4078 { | |
| 4079 ALmixer_SetError("%s", | |
|
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|
4080 alGetString(error) ); |
|
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|
4081 retval = AL_FALSE; |
| 0 | 4082 } |
| 4083 } | |
| 4084 } | |
| 4085 } | |
| 4086 return retval; | |
| 4087 } | |
| 4088 | |
|
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4089 static ALboolean Internal_SetMinVolumeSource(ALuint source, ALfloat volume) |
|
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|
4090 { |
|
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|
4091 ALint channel; |
| 0 | 4092 if(0 == source) |
| 4093 { | |
| 4094 return Internal_SetMinVolumeChannel(-1, volume); | |
| 4095 } | |
| 4096 | |
| 4097 channel = Internal_GetChannel(source); | |
| 4098 if(-1 == channel) | |
| 4099 { | |
| 4100 ALmixer_SetError("Cannot SetMaxVolume: %s", ALmixer_GetError()); | |
|
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|
4101 return AL_FALSE; |
| 0 | 4102 } |
| 4103 return Internal_SetMinVolumeChannel(channel, volume); | |
| 4104 } | |
| 4105 | |
|
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|
4106 static ALfloat Internal_GetMinVolumeChannel(ALint channel) |
| 0 | 4107 { |
| 4108 /* | |
| 4109 ALfloat value; | |
| 4110 ALenum error; | |
| 4111 */ | |
| 4112 ALfloat running_total = 0.0f; | |
| 4113 ALfloat retval = 0.0f; | |
| 4114 | |
| 4115 if(channel >= Number_of_Channels_global) | |
| 4116 { | |
| 4117 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); | |
| 4118 return -1.0f; | |
| 4119 } | |
| 4120 | |
| 4121 if(channel >= 0) | |
| 4122 { | |
| 4123 /* | |
| 4124 alGetSourcef(ALmixer_Channel_List[channel].alsource, | |
| 4125 AL_GAIN, &value); | |
| 4126 if((error = alGetError()) != AL_NO_ERROR) | |
| 4127 { | |
| 4128 ALmixer_SetError("%s", | |
|
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|
4129 alGetString(error) ); |
| 0 | 4130 retval = -1.0f; |
| 4131 } | |
| 4132 else | |
| 4133 { | |
| 4134 retval = value; | |
| 4135 } | |
| 4136 */ | |
| 4137 retval = ALmixer_Channel_List[channel].min_volume; | |
| 4138 | |
| 4139 } | |
| 4140 else | |
| 4141 { | |
|
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|
4142 ALint i; |
| 0 | 4143 for(i=0; i<Number_of_Channels_global; i++) |
| 4144 { | |
| 4145 /* | |
| 4146 alGetSourcef(ALmixer_Channel_List[i].alsource, | |
| 4147 AL_GAIN, &value); | |
| 4148 if((error = alGetError()) != AL_NO_ERROR) | |
| 4149 { | |
| 4150 ALmixer_SetError("%s", | |
|
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|
4151 alGetString(error) ); |
| 0 | 4152 retval = -1; |
| 4153 } | |
| 4154 else | |
| 4155 { | |
| 4156 running_total += value; | |
| 4157 } | |
| 4158 */ | |
| 4159 running_total += ALmixer_Channel_List[i].min_volume; | |
| 4160 } | |
| 4161 if(0 == Number_of_Channels_global) | |
| 4162 { | |
| 4163 ALmixer_SetError("No channels are allocated"); | |
| 4164 retval = -1.0f; | |
| 4165 } | |
| 4166 else | |
| 4167 { | |
| 4168 retval = running_total / Number_of_Channels_global; | |
| 4169 } | |
| 4170 } | |
| 4171 return retval; | |
| 4172 } | |
| 4173 | |
| 4174 static ALfloat Internal_GetMinVolumeSource(ALuint source) | |
| 4175 { | |
|
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|
4176 ALint channel; |
| 0 | 4177 if(0 == source) |
| 4178 { | |
| 4179 return Internal_GetMinVolumeChannel(-1); | |
| 4180 } | |
| 4181 | |
| 4182 channel = Internal_GetChannel(source); | |
| 4183 if(-1 == channel) | |
| 4184 { | |
| 4185 ALmixer_SetError("Cannot GetVolume: %s", ALmixer_GetError()); | |
| 4186 return -1.0f; | |
| 4187 } | |
| 4188 | |
| 4189 return Internal_GetMinVolumeChannel(channel); | |
| 4190 } | |
| 4191 | |
| 4192 | |
| 4193 /* Changes the listener volume */ | |
|
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|
4194 static ALboolean Internal_SetMasterVolume(ALfloat volume) |
| 0 | 4195 { |
| 4196 ALenum error; | |
| 4197 alListenerf(AL_GAIN, volume); | |
| 4198 if((error = alGetError()) != AL_NO_ERROR) | |
| 4199 { | |
| 4200 ALmixer_SetError("%s", | |
|
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|
4201 alGetString(error) ); |
|
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|
4202 return AL_FALSE; |
|
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|
4203 } |
|
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|
4204 return AL_TRUE; |
| 0 | 4205 } |
| 4206 | |
| 4207 static ALfloat Internal_GetMasterVolume() | |
| 4208 { | |
| 4209 ALenum error; | |
| 4210 ALfloat volume; | |
| 4211 alGetListenerf(AL_GAIN, &volume); | |
| 4212 if((error = alGetError()) != AL_NO_ERROR) | |
| 4213 { | |
| 4214 ALmixer_SetError("%s", | |
|
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|
4215 alGetString(error) ); |
| 0 | 4216 return -1.0f; |
| 4217 } | |
| 4218 return volume; | |
| 4219 } | |
| 4220 | |
| 4221 | |
| 4222 | |
| 4223 | |
| 4224 /* Will fade out currently playing channels. | |
| 4225 * It starts at the current volume level and goes down */ | |
|
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|
4226 static ALint Internal_ExpireChannel(ALint channel, ALint ticks) |
|
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4227 { |
|
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|
4228 ALuint current_time = ALmixer_GetTicks(); |
|
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|
4229 ALuint counter = 0; |
| 0 | 4230 |
| 4231 /* We can't accept 0 as a value because of div-by-zero. | |
| 4232 * If zero, just call Halt at normal | |
| 4233 * volume | |
| 4234 */ | |
| 4235 if(0 == ticks) | |
| 4236 { | |
|
10
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|
4237 return Internal_HaltChannel(channel, AL_FALSE); |
| 0 | 4238 } |
| 4239 if(ticks < -1) | |
| 4240 { | |
| 4241 ticks = -1; | |
| 4242 } | |
| 4243 | |
| 4244 | |
| 4245 if(channel >= Number_of_Channels_global) | |
| 4246 { | |
| 4247 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); | |
| 4248 return -1; | |
| 4249 } | |
| 4250 | |
| 4251 if(channel >= 0) | |
| 4252 { | |
| 4253 if(ALmixer_Channel_List[channel].channel_in_use) | |
| 4254 { | |
| 4255 /* Set expire start time */ | |
| 4256 ALmixer_Channel_List[channel].start_time = current_time; | |
| 4257 /* Set the expire ticks */ | |
| 4258 ALmixer_Channel_List[channel].expire_ticks = ticks; | |
| 4259 | |
| 4260 counter++; | |
| 4261 } | |
| 4262 } | |
| 4263 /* Else need to fade out all channels */ | |
| 4264 else | |
| 4265 { | |
|
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|
4266 ALint i; |
| 0 | 4267 for(i=0; i<Number_of_Channels_global; i++) |
| 4268 { | |
| 4269 if(ALmixer_Channel_List[i].channel_in_use) | |
| 4270 { | |
| 4271 /* Set expire start time */ | |
| 4272 ALmixer_Channel_List[i].start_time = current_time; | |
| 4273 /* Set the expire ticks */ | |
| 4274 ALmixer_Channel_List[i].expire_ticks = ticks; | |
| 4275 | |
| 4276 counter++; | |
| 4277 } | |
| 4278 } /* End for loop */ | |
| 4279 } | |
| 4280 return counter; | |
| 4281 } | |
| 4282 | |
| 4283 | |
|
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|
4284 static ALint Internal_ExpireSource(ALuint source, ALint ticks) |
|
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|
4285 { |
|
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|
4286 ALint channel; |
| 0 | 4287 if(0 == source) |
| 4288 { | |
| 4289 return Internal_ExpireChannel(-1, ticks); | |
| 4290 } | |
| 4291 | |
| 4292 channel = Internal_GetChannel(source); | |
| 4293 if(-1 == channel) | |
| 4294 { | |
| 4295 ALmixer_SetError("Cannot Expire source: %s", ALmixer_GetError()); | |
| 4296 return -1; | |
| 4297 } | |
| 4298 return Internal_ExpireChannel(channel, ticks); | |
| 4299 } | |
| 4300 | |
| 4301 | |
|
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|
4302 static ALint Internal_QueryChannel(ALint channel) |
|
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|
4303 { |
|
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|
4304 ALint i; |
|
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|
4305 ALint counter = 0; |
| 0 | 4306 if(channel >= Number_of_Channels_global) |
| 4307 { | |
| 4308 ALmixer_SetError("Invalid channel: %d", channel); | |
| 4309 return -1; | |
| 4310 } | |
| 4311 | |
| 4312 if(channel >= 0) | |
| 4313 { | |
| 4314 return ALmixer_Channel_List[channel].channel_in_use; | |
| 4315 } | |
| 4316 | |
| 4317 /* Else, return the number of channels in use */ | |
| 4318 for(i=0; i<Number_of_Channels_global; i++) | |
| 4319 { | |
| 4320 if(ALmixer_Channel_List[i].channel_in_use) | |
| 4321 { | |
| 4322 counter++; | |
| 4323 } | |
| 4324 } | |
| 4325 return counter; | |
| 4326 } | |
| 4327 | |
| 4328 | |
|
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|
4329 static ALint Internal_QuerySource(ALuint source) |
|
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|
4330 { |
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|
4331 ALint channel; |
| 0 | 4332 if(0 == source) |
| 4333 { | |
| 4334 return Internal_QueryChannel(-1); | |
| 4335 } | |
| 4336 | |
| 4337 channel = Internal_GetChannel(source); | |
| 4338 if(-1 == channel) | |
| 4339 { | |
| 4340 ALmixer_SetError("Cannot query source: %s", ALmixer_GetError()); | |
| 4341 return -1; | |
| 4342 } | |
| 4343 | |
| 4344 return Internal_QueryChannel(channel); | |
| 4345 } | |
| 4346 | |
| 4347 | |
|
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|
4348 static ALuint Internal_CountUnreservedUsedChannels() |
|
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|
4349 { |
|
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|
4350 ALint i; |
|
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|
4351 ALuint counter = 0; |
| 0 | 4352 |
| 4353 | |
| 4354 /* Else, return the number of channels in use */ | |
| 4355 for(i=Number_of_Reserve_Channels_global; i<Number_of_Channels_global; i++) | |
| 4356 { | |
| 4357 if(ALmixer_Channel_List[i].channel_in_use) | |
| 4358 { | |
| 4359 counter++; | |
| 4360 } | |
| 4361 } | |
| 4362 return counter; | |
| 4363 } | |
| 4364 | |
|
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|
4365 static ALuint Internal_CountUnreservedFreeChannels() |
|
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|
4366 { |
|
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|
4367 ALint i; |
|
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|
4368 ALuint counter = 0; |
| 0 | 4369 |
| 4370 | |
| 4371 /* Else, return the number of channels in use */ | |
| 4372 for(i=Number_of_Reserve_Channels_global; i<Number_of_Channels_global; i++) | |
| 4373 { | |
| 4374 if( ! ALmixer_Channel_List[i].channel_in_use) | |
| 4375 { | |
| 4376 counter++; | |
| 4377 } | |
| 4378 } | |
| 4379 return counter; | |
| 4380 } | |
| 4381 | |
|
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|
4382 static ALuint Internal_CountAllUsedChannels() |
|
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|
4383 { |
|
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|
4384 ALint i; |
|
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|
4385 ALuint counter = 0; |
| 0 | 4386 |
| 4387 | |
| 4388 /* Else, return the number of channels in use */ | |
| 4389 for(i=0; i<Number_of_Channels_global; i++) | |
| 4390 { | |
| 4391 if(ALmixer_Channel_List[i].channel_in_use) | |
| 4392 { | |
| 4393 counter++; | |
| 4394 } | |
| 4395 } | |
| 4396 return counter; | |
| 4397 } | |
| 4398 | |
|
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|
4399 static ALuint Internal_CountAllFreeChannels() |
|
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|
4400 { |
|
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|
4401 ALint i; |
|
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|
4402 ALuint counter = 0; |
| 0 | 4403 |
| 4404 | |
| 4405 /* Else, return the number of channels in use */ | |
| 4406 for(i=0; i<Number_of_Channels_global; i++) | |
| 4407 { | |
| 4408 if( ! ALmixer_Channel_List[i].channel_in_use) | |
| 4409 { | |
| 4410 counter++; | |
| 4411 } | |
| 4412 } | |
| 4413 return counter; | |
| 4414 } | |
| 4415 | |
| 4416 | |
|
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|
4417 static ALint Internal_PlayingChannel(ALint channel) |
|
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|
4418 { |
|
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|
4419 ALint i; |
|
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|
4420 ALint counter = 0; |
| 0 | 4421 ALint state; |
| 4422 | |
| 4423 if(channel >= Number_of_Channels_global) | |
| 4424 { | |
| 4425 ALmixer_SetError("Invalid channel: %d", channel); | |
| 4426 return -1; | |
| 4427 } | |
| 4428 | |
| 4429 if(channel >= 0) | |
| 4430 { | |
| 4431 if(ALmixer_Channel_List[channel].channel_in_use) | |
| 4432 { | |
| 4433 alGetSourcei( | |
| 4434 ALmixer_Channel_List[channel].alsource, | |
| 4435 AL_SOURCE_STATE, &state | |
| 4436 ); | |
| 4437 if(AL_PLAYING == state) | |
| 4438 { | |
| 4439 return 1; | |
| 4440 } | |
| 4441 } | |
| 4442 return 0; | |
| 4443 } | |
| 4444 | |
| 4445 /* Else, return the number of channels in use */ | |
| 4446 for(i=0; i<Number_of_Channels_global; i++) | |
| 4447 { | |
| 4448 if(ALmixer_Channel_List[i].channel_in_use) | |
| 4449 { | |
| 4450 alGetSourcei( | |
| 4451 ALmixer_Channel_List[i].alsource, | |
| 4452 AL_SOURCE_STATE, &state | |
| 4453 ); | |
| 4454 if(AL_PLAYING == state) | |
| 4455 { | |
| 4456 counter++; | |
| 4457 } | |
| 4458 } | |
| 4459 } | |
| 4460 return counter; | |
| 4461 } | |
| 4462 | |
| 4463 | |
|
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|
4464 static ALint Internal_PlayingSource(ALuint source) |
|
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|
4465 { |
|
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|
4466 ALint channel; |
| 0 | 4467 if(0 == source) |
| 4468 { | |
| 4469 return Internal_PlayingChannel(-1); | |
| 4470 } | |
| 4471 | |
| 4472 channel = Internal_GetChannel(source); | |
| 4473 if(-1 == channel) | |
| 4474 { | |
| 4475 ALmixer_SetError("Cannot query source: %s", ALmixer_GetError()); | |
| 4476 return -1; | |
| 4477 } | |
| 4478 | |
| 4479 return Internal_PlayingChannel(channel); | |
| 4480 } | |
| 4481 | |
| 4482 | |
|
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|
4483 static ALint Internal_PausedChannel(ALint channel) |
|
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|
4484 { |
|
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|
4485 ALint i; |
|
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|
4486 ALint counter = 0; |
| 0 | 4487 ALint state; |
| 4488 | |
| 4489 if(channel >= Number_of_Channels_global) | |
| 4490 { | |
| 4491 ALmixer_SetError("Invalid channel: %d", channel); | |
| 4492 return -1; | |
| 4493 } | |
| 4494 | |
| 4495 if(channel >= 0) | |
| 4496 { | |
| 4497 if(ALmixer_Channel_List[channel].channel_in_use) | |
| 4498 { | |
| 4499 alGetSourcei( | |
| 4500 ALmixer_Channel_List[channel].alsource, | |
| 4501 AL_SOURCE_STATE, &state | |
| 4502 ); | |
| 4503 if(AL_PAUSED == state) | |
| 4504 { | |
| 4505 return 1; | |
| 4506 } | |
| 4507 } | |
| 4508 return 0; | |
| 4509 } | |
| 4510 | |
| 4511 /* Else, return the number of channels in use */ | |
| 4512 for(i=0; i<Number_of_Channels_global; i++) | |
| 4513 { | |
| 4514 if(ALmixer_Channel_List[i].channel_in_use) | |
| 4515 { | |
| 4516 alGetSourcei( | |
| 4517 ALmixer_Channel_List[i].alsource, | |
| 4518 AL_SOURCE_STATE, &state | |
| 4519 ); | |
| 4520 if(AL_PAUSED == state) | |
| 4521 { | |
| 4522 counter++; | |
| 4523 } | |
| 4524 } | |
| 4525 } | |
| 4526 return counter; | |
| 4527 } | |
| 4528 | |
| 4529 | |
|
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|
4530 static ALint Internal_PausedSource(ALuint source) |
|
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|
4531 { |
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|
4532 ALint channel; |
| 0 | 4533 if(0 == source) |
| 4534 { | |
| 4535 return Internal_PausedChannel(-1); | |
| 4536 } | |
| 4537 | |
| 4538 channel = Internal_GetChannel(source); | |
| 4539 if(-1 == channel) | |
| 4540 { | |
| 4541 ALmixer_SetError("Cannot query source: %s", ALmixer_GetError()); | |
| 4542 return -1; | |
| 4543 } | |
| 4544 | |
| 4545 return Internal_PausedChannel(channel); | |
| 4546 } | |
| 4547 | |
| 4548 | |
| 4549 | |
| 4550 | |
| 4551 | |
| 4552 | |
| 4553 /* Private function for Updating ALmixer. | |
| 4554 * This is a very big and ugly function. | |
| 4555 * It should return the number of buffers that were | |
| 4556 * queued during the call. The value might be | |
| 4557 * used to guage how long you might wait to | |
| 4558 * call the next update loop in case you are worried | |
| 4559 * about preserving CPU cycles. The idea is that | |
| 4560 * when a buffer is queued, there was probably some | |
| 4561 * CPU intensive looping which took awhile. | |
| 4562 * It's mainly provided as a convenience. | |
|
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|
4563 * Timing the call with ALmixer_GetTicks() would produce |
| 0 | 4564 * more accurate information. |
| 4565 * Returns a negative value if there was an error, | |
| 4566 * the value being the number of errors. | |
| 4567 */ | |
|
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|
4568 static ALint Update_ALmixer(void* data) |
|
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|
4569 { |
|
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|
4570 ALint retval = 0; |
|
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|
4571 ALint error_flag = 0; |
| 0 | 4572 ALenum error; |
| 4573 ALint state; | |
|
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|
4574 ALint i=0; |
|
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|
4575 |
|
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|
4576 #ifdef ENABLE_ALMIXER_THREADS |
|
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|
4577 SDL_LockMutex(s_simpleLock); |
|
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4578 #endif |
| 0 | 4579 if(0 == ALmixer_Initialized) |
| 4580 { | |
|
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|
4581 #ifdef ENABLE_ALMIXER_THREADS |
|
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|
4582 SDL_UnlockMutex(s_simpleLock); |
|
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|
4583 #endif |
| 0 | 4584 return 0; |
| 4585 } | |
| 4586 | |
| 4587 /* Check the quick flag to see if anything needs updating */ | |
| 4588 /* If anything is playing, then we have to do work */ | |
| 4589 if( 0 == Is_Playing_global) | |
| 4590 { | |
|
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|
4591 #ifdef ENABLE_ALMIXER_THREADS |
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|
4592 SDL_UnlockMutex(s_simpleLock); |
|
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|
4593 #endif |
| 0 | 4594 return 0; |
| 4595 } | |
| 4596 /* Clear error */ | |
| 4597 if((error = alGetError()) != AL_NO_ERROR) | |
| 4598 { | |
| 4599 fprintf(stderr, "08Testing errpr before unqueue because getting stuff, for OS X this is expected: %s\n", | |
|
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|
4600 alGetString(error)); |
| 0 | 4601 } |
| 4602 alGetError(); | |
| 4603 | |
| 4604 for(i=0; i<Number_of_Channels_global; i++) | |
| 4605 { | |
| 4606 if( ALmixer_Channel_List[i].channel_in_use ) | |
| 4607 { | |
| 4608 | |
| 4609 /* For simplicity, before we do anything else, | |
| 4610 * we can check the timeout and fading values | |
| 4611 * and do the appropriate things | |
| 4612 */ | |
|
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|
4613 ALuint current_time = ALmixer_GetTicks(); |
| 0 | 4614 |
| 4615 /* Check to see if we need to halt due to Timed play */ | |
| 4616 if(ALmixer_Channel_List[i].expire_ticks != -1) | |
| 4617 { | |
|
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|
4618 ALuint target_time = (ALuint)ALmixer_Channel_List[i].expire_ticks |
| 0 | 4619 + ALmixer_Channel_List[i].start_time; |
| 4620 alGetSourcei(ALmixer_Channel_List[i].alsource, | |
| 4621 AL_SOURCE_STATE, &state); | |
| 4622 if((error = alGetError()) != AL_NO_ERROR) | |
| 4623 { | |
| 4624 fprintf(stderr, "06Testing errpr before unqueue because getting stuff, for OS X this is expected: %s\n", | |
|
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|
4625 alGetString(error)); |
| 0 | 4626 } |
| 4627 | |
| 4628 /* Check the time, and also make sure that it is not | |
| 4629 * paused (if paused, we don't want to make the | |
| 4630 * evaluation because when resumed, we will adjust | |
| 4631 * the times to compensate for the pause). | |
| 4632 */ | |
| 4633 if( (current_time >= target_time) | |
| 4634 && (state != AL_PAUSED) ) | |
| 4635 { | |
| 4636 /* Stop the playback */ | |
|
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6
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changeset
|
4637 Internal_HaltChannel(i, AL_FALSE); |
| 0 | 4638 if((error = alGetError()) != AL_NO_ERROR) |
| 4639 { | |
| 4640 fprintf(stderr, "07Testing errpr before unqueue because getting stuff, for OS X this is expected: %s\n", | |
|
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|
4641 alGetString(error)); |
| 0 | 4642 } |
| 4643 | |
| 4644 /* Everything should be done so go on to the next loop */ | |
| 4645 continue; | |
| 4646 } | |
| 4647 } /* End if time expired check */ | |
| 4648 | |
| 4649 /* Check to see if we need to adjust the volume for fading */ | |
| 4650 if( ALmixer_Channel_List[i].fade_enabled ) | |
| 4651 { | |
|
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|
4652 ALuint target_time = ALmixer_Channel_List[i].fade_expire_ticks |
| 0 | 4653 + ALmixer_Channel_List[i].fade_start_time; |
| 4654 alGetSourcei(ALmixer_Channel_List[i].alsource, | |
| 4655 AL_SOURCE_STATE, &state); | |
| 4656 if((error = alGetError()) != AL_NO_ERROR) | |
| 4657 { | |
| 4658 fprintf(stderr, "05Testing errpr before unqueue because getting stuff, for OS X this is expected: %s\n", | |
|
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|
4659 alGetString(error)); |
| 0 | 4660 } |
| 4661 | |
| 4662 /* Check the time, and also make sure that it is not | |
| 4663 * paused (if paused, we don't want to make the | |
| 4664 * evaluation because when resumed, we will adjust | |
| 4665 * the times to compensate for the pause). | |
| 4666 */ | |
| 4667 if(state != AL_PAUSED) | |
| 4668 { | |
| 4669 ALfloat t; | |
|
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|
4670 ALuint delta_time; |
| 0 | 4671 ALfloat current_volume; |
| 4672 if(current_time >= target_time) | |
| 4673 { | |
| 4674 /* Need to constrain value to the end time | |
| 4675 * (can't go pass the value for calculations) | |
| 4676 */ | |
| 4677 current_time = target_time; | |
| 4678 /* We can disable the fade flag now */ | |
| 4679 ALmixer_Channel_List[i].fade_enabled = 0; | |
| 4680 } | |
| 4681 /* Use the linear interpolation formula: | |
| 4682 * X = (1-t)x0 + tx1 | |
| 4683 * where x0 would be the start value | |
| 4684 * and x1 is the final value | |
| 4685 * and t is delta_time*inv_time (adjusts 0 <= time <= 1) | |
| 4686 * delta_time = current_time-start_time | |
| 4687 * inv_time = 1/ (end_time-start_time) | |
| 4688 * so t = current_time-start_time / (end_time-start_time) | |
| 4689 * | |
| 4690 */ | |
| 4691 delta_time = current_time - ALmixer_Channel_List[i].fade_start_time; | |
| 4692 t = (ALfloat) delta_time * ALmixer_Channel_List[i].fade_inv_time; | |
| 4693 current_volume = (1.0f-t) * ALmixer_Channel_List[i].fade_start_volume | |
| 4694 + t * ALmixer_Channel_List[i].fade_end_volume; | |
|
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|
4695 fprintf(stderr, "start_vol=%f, end_vol:%f, current_volume: %f\n", ALmixer_Channel_List[i].fade_start_volume, ALmixer_Channel_List[i].fade_end_volume, current_volume); |
| 0 | 4696 |
| 4697 /* Set the volume */ | |
| 4698 alSourcef(ALmixer_Channel_List[i].alsource, | |
|
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|
4699 AL_GAIN, current_volume); |
| 0 | 4700 if((error = alGetError()) != AL_NO_ERROR) |
| 4701 { | |
| 4702 fprintf(stderr, "04Testing errpr before unqueue because getting stuff, for OS X this is expected: %s\n", | |
|
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|
4703 alGetString(error)); |
| 0 | 4704 } |
| 4705 | |
| 4706 /* | |
| 4707 fprintf(stderr, "Current time =%d\n", current_time); | |
| 4708 fprintf(stderr, "Current vol=%f on channel %d\n", current_volume, i); | |
| 4709 */ | |
| 4710 } /* End if not PAUSED */ | |
| 4711 } /* End if fade_enabled */ | |
| 4712 | |
| 4713 | |
| 4714 /* Okay, now that the time expired and fading stuff | |
| 4715 * is done, do the rest of the hard stuff | |
| 4716 */ | |
| 4717 | |
| 4718 | |
| 4719 /* For predecoded, check to see if done */ | |
| 4720 if( ALmixer_Channel_List[i].almixer_data->decoded_all ) | |
| 4721 { | |
| 4722 | |
| 4723 #if 0 | |
| 4724 /********* Remove this **********/ | |
| 4725 ALint buffers_processed; | |
| 4726 ALint buffers_still_queued; | |
| 4727 fprintf(stderr, "For Predecoded\n"); | |
| 4728 | |
| 4729 alGetSourcei( | |
| 4730 ALmixer_Channel_List[i].alsource, | |
| 4731 AL_SOURCE_STATE, &state | |
| 4732 ); | |
| 4733 switch(state) { | |
| 4734 case AL_PLAYING: | |
| 4735 fprintf(stderr, "Channel '%d' is PLAYING\n", i); | |
| 4736 break; | |
| 4737 case AL_PAUSED: | |
| 4738 fprintf(stderr, "Channel '%d' is PAUSED\n",i); | |
| 4739 break; | |
| 4740 case AL_STOPPED: | |
| 4741 fprintf(stderr, "Channel '%d' is STOPPED\n",i); | |
| 4742 break; | |
| 4743 case AL_INITIAL: | |
| 4744 fprintf(stderr, "Channel '%d' is INITIAL\n",i); | |
| 4745 break; | |
| 4746 default: | |
| 4747 fprintf(stderr, "Channel '%d' is UNKNOWN\n",i); | |
| 4748 break; | |
| 4749 } | |
| 4750 | |
| 4751 alGetSourcei( | |
| 4752 ALmixer_Channel_List[i].alsource, | |
| 4753 AL_BUFFERS_PROCESSED, &buffers_processed | |
| 4754 ); | |
| 4755 fprintf(stderr, "Buffers processed = %d\n", buffers_processed); | |
| 4756 | |
| 4757 alGetSourcei( | |
| 4758 ALmixer_Channel_List[i].alsource, | |
| 4759 AL_BUFFERS_QUEUED, &buffers_still_queued | |
| 4760 ); | |
| 4761 | |
| 4762 /******** END REMOVE *******/ | |
| 4763 #endif | |
| 4764 /* FIXME: Ugh! Somewhere an alError is being thrown ("Invalid Enum Value"), but I can't | |
| 4765 * find it. It only seems to be thrown for OS X. I placed error messages after every al* | |
| 4766 * command I could find in the above loops, but the error doesn't seem to show | |
| 4767 * up until around here. I mistook it for a get queued buffers | |
| 4768 * error in OS X. I don't think there's an error down there. | |
| 4769 * For now, I'm clearing the error here. | |
| 4770 */ | |
| 4771 | |
| 4772 if((error = alGetError()) != AL_NO_ERROR) | |
| 4773 { | |
| 4774 fprintf(stderr, "03Testing errpr before unqueue because getting stuff, for OS X this is expected: %s\n", | |
|
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|
4775 alGetString(error)); |
| 0 | 4776 } |
| 4777 | |
| 4778 | |
| 4779 alGetSourcei( | |
| 4780 ALmixer_Channel_List[i].alsource, | |
| 4781 AL_SOURCE_STATE, &state | |
| 4782 ); | |
| 4783 if((error = alGetError()) != AL_NO_ERROR) | |
| 4784 { | |
| 4785 fprintf(stderr, "02Testing errpr before unqueue because getting stuff, for OS X this is expected: %s\n", | |
|
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|
4786 alGetString(error)); |
| 0 | 4787 } |
| 4788 | |
| 4789 | |
| 4790 if(AL_STOPPED == state) | |
| 4791 { | |
| 4792 /* Playback has ended. | |
| 4793 * Loop if necessary, or launch callback | |
| 4794 * and clear channel (or clear channel and | |
| 4795 * then launch callback?) | |
| 4796 */ | |
| 4797 | |
| 4798 | |
| 4799 /* Need to check for loops */ | |
| 4800 if(ALmixer_Channel_List[i].loops != 0) | |
| 4801 { | |
| 4802 /* Corner Case: If the buffer has | |
| 4803 * been modified using Seek, | |
| 4804 * the loop will start at the seek | |
| 4805 * position. | |
| 4806 */ | |
| 4807 if(ALmixer_Channel_List[i].loops != -1) | |
| 4808 { | |
| 4809 ALmixer_Channel_List[i].loops--; | |
| 4810 } | |
| 4811 alSourcePlay(ALmixer_Channel_List[i].alsource); | |
| 4812 if((error = alGetError()) != AL_NO_ERROR) | |
| 4813 { | |
| 4814 fprintf(stderr, "50Testing error: %s\n", | |
|
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|
4815 alGetString(error)); |
| 0 | 4816 } |
| 4817 continue; | |
| 4818 } | |
| 4819 /* No loops. End play. */ | |
| 4820 else | |
| 4821 { | |
| 4822 /* Problem: It seems that when mixing | |
| 4823 * streamed and predecoded sources, | |
| 4824 * the previous instance lingers, | |
| 4825 * so we need to force remove | |
| 4826 * the data from the source. | |
| 4827 * The sharing problem | |
| 4828 * occurs when a previous predecoded buffer is played on | |
| 4829 * a source, and then a streamed source is played later | |
| 4830 * on that same source. OpenAL isn't consistently | |
| 4831 * removing the previous buffer so both get played. | |
| 4832 * (Different dists seem to have different quirks. | |
| 4833 * The problem might lead to crashes in the worst case.) | |
| 4834 */ | |
| 4835 /* Additional problem: There is another | |
| 4836 * inconsistency among OpenAL distributions. | |
| 4837 * Both Loki and Creative Windows seem to keep | |
| 4838 * the buffer queued which requires removing. | |
| 4839 * But the Creative Macintosh version does | |
| 4840 * not have any buffer queued after play | |
| 4841 * and it returns the error: Invalid Enum Value | |
| 4842 * if I try to unqueue it. | |
| 4843 * So I'm going to put in a check to see if I | |
| 4844 * can detect any buffers queued first | |
| 4845 * and then unqueue them if I can see them. | |
| 4846 * Additional note: The new CoreAudio based | |
| 4847 * implementation leaves it's buffer queued | |
| 4848 * like Loki and Creative Windows. But | |
| 4849 * considering all the problems I'm having | |
| 4850 * with the different distributions, this | |
| 4851 * check seems reasonable. | |
| 4852 */ | |
| 4853 ALint buffers_still_queued; | |
| 4854 if((error = alGetError()) != AL_NO_ERROR) | |
| 4855 { | |
| 4856 fprintf(stderr, "01Testing errpr before unqueue because getting stuff, for OS X this is expected: %s\n", | |
|
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|
4857 alGetString(error)); |
| 0 | 4858 } |
| 4859 | |
| 4860 alGetSourcei( | |
| 4861 ALmixer_Channel_List[i].alsource, | |
| 4862 AL_BUFFERS_QUEUED, &buffers_still_queued | |
| 4863 ); | |
| 4864 if((error = alGetError()) != AL_NO_ERROR) | |
| 4865 { | |
| 4866 fprintf(stderr, "Error with unqueue, for OS X this is expected: %s\n", | |
|
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|
4867 alGetString(error)); |
| 0 | 4868 ALmixer_SetError("Failed detecting unqueued predecoded buffer (expected with OS X): %s", |
|
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|
4869 alGetString(error) ); |
| 0 | 4870 error_flag--; |
| 4871 } | |
| 4872 if(buffers_still_queued > 0) | |
| 4873 { | |
| 1 | 4874 |
| 4875 #if 0 /* This triggers an error in OS X Core Audio. */ | |
| 4876 alSourceUnqueueBuffers( | |
| 4877 ALmixer_Channel_List[i].alsource, | |
| 4878 1, | |
| 4879 ALmixer_Channel_List[i].almixer_data->buffer | |
| 4880 ); | |
| 4881 #else | |
| 4882 /* fprintf(stderr, "In the Bob Aron section...about to clear source\n"); | |
| 0 | 4883 PrintQueueStatus(ALmixer_Channel_List[i].alsource); |
| 1 | 4884 */ |
| 0 | 4885 /* Rather than force unqueuing the buffer, let's see if |
| 4886 * setting the buffer to none works (the OpenAL 1.0 | |
| 4887 * Reference Annotation suggests this should work). | |
| 4888 */ | |
| 4889 alSourcei(ALmixer_Channel_List[i].alsource, | |
| 4890 AL_BUFFER, AL_NONE); | |
| 4891 /* | |
| 4892 PrintQueueStatus(ALmixer_Channel_List[i].alsource); | |
| 4893 */ | |
| 1 | 4894 #endif |
| 0 | 4895 if((error = alGetError()) != AL_NO_ERROR) |
| 4896 { | |
| 4897 fprintf(stderr, "Error with unqueue, after alSourceUnqueueBuffers, buffers_still_queued=%d, error is: %s", buffers_still_queued, | |
|
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|
4898 alGetString(error)); |
| 0 | 4899 ALmixer_SetError("Predecoded Unqueue buffer failed: %s", |
|
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|
4900 alGetString(error) ); |
| 0 | 4901 error_flag--; |
| 4902 } | |
| 4903 | |
| 4904 } | |
| 4905 | |
|
10
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|
4906 /* Launch callback */ |
|
c808684660a7
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6
diff
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|
4907 Invoke_Channel_Done_Callback(i, AL_TRUE); |
|
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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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|
4908 |
| 0 | 4909 Clean_Channel(i); |
| 4910 /* Subtract counter */ | |
| 4911 Is_Playing_global--; | |
| 4912 | |
| 4913 | |
| 4914 /* We're done for this loop. | |
| 4915 * Go to next channel | |
| 4916 */ | |
| 4917 continue; | |
| 4918 } | |
| 4919 continue; | |
| 4920 } | |
| 4921 } /* End if decoded_all */ | |
| 4922 /* For streamed */ | |
| 4923 else | |
| 4924 { | |
| 4925 ALint buffers_processed; | |
| 4926 ALint buffers_still_queued; | |
| 4927 ALint current_buffer_id; | |
| 4928 | |
| 4929 ALuint unqueued_buffer_id; | |
| 4930 #if 0 | |
| 4931 /********* Remove this **********/ | |
| 4932 fprintf(stderr, "For Streamed\n"); | |
| 4933 | |
| 4934 alGetSourcei( | |
| 4935 ALmixer_Channel_List[i].alsource, | |
| 4936 AL_SOURCE_STATE, &state | |
| 4937 ); | |
| 4938 switch(state) { | |
| 4939 case AL_PLAYING: | |
| 4940 fprintf(stderr, "Channel '%d' is PLAYING\n", i); | |
| 4941 break; | |
| 4942 case AL_PAUSED: | |
| 4943 fprintf(stderr, "Channel '%d' is PAUSED\n",i); | |
| 4944 break; | |
| 4945 case AL_STOPPED: | |
| 4946 fprintf(stderr, "Channel '%d' is STOPPED\n",i); | |
| 4947 break; | |
| 4948 case AL_INITIAL: | |
| 4949 fprintf(stderr, "Channel '%d' is INITIAL\n",i); | |
| 4950 break; | |
| 4951 default: | |
| 4952 fprintf(stderr, "Channel '%d' is UNKNOWN\n",i); | |
| 4953 break; | |
| 4954 } | |
| 4955 /******** END REMOVE *******/ | |
| 4956 #endif | |
| 4957 /* Get the number of buffers still queued */ | |
| 4958 alGetSourcei( | |
| 4959 ALmixer_Channel_List[i].alsource, | |
| 4960 AL_BUFFERS_QUEUED, &buffers_still_queued | |
| 4961 ); | |
| 4962 if((error = alGetError()) != AL_NO_ERROR) | |
| 4963 { | |
| 4964 fprintf(stderr, "51Testing error: %s\n", | |
|
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|
4965 alGetString(error)); |
| 0 | 4966 } |
| 4967 /* Get the number of buffers processed | |
| 4968 * so we know if we need to refill | |
| 4969 */ | |
|
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diff
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|
4970 /* WARNING: It looks like Snow Leopard some times crashes on this call under x86_64 |
|
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|
4971 * typically when I suffer a lot of buffer underruns. |
|
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diff
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|
4972 */ |
|
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diff
changeset
|
4973 // fprintf(stderr, "calling AL_BUFFERS_PROCESSED on source:%d", ALmixer_Channel_List[i].alsource); |
| 0 | 4974 alGetSourcei( |
| 4975 ALmixer_Channel_List[i].alsource, | |
| 4976 AL_BUFFERS_PROCESSED, &buffers_processed | |
| 4977 ); | |
| 4978 if((error = alGetError()) != AL_NO_ERROR) | |
| 4979 { | |
| 4980 fprintf(stderr, "52Testing error: %s\n", | |
|
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4981 alGetString(error)); |
| 0 | 4982 } |
|
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4983 // fprintf(stderr, "finished AL_BUFFERS_PROCESSED, buffers_processed=%d", buffers_processed); |
| 0 | 4984 |
| 4985 /* WTF!!! The Nvidia distribution is failing on the alGetSourcei(source, AL_BUFFER, buf_id) call. | |
| 4986 * I need this call to figure out which buffer OpenAL is currently playing. | |
| 4987 * It keeps returning an "Invalid Enum" error. | |
| 4988 * This is totally inane! It's a basic query. | |
| 4989 * By the spec, this functionality is not explicitly defined so Nvidia refuses to | |
| 4990 * fix this behavior, even though all other distributions work fine with this. | |
| 4991 * The only workaround for this is for | |
| 4992 * a significant rewrite of my code which requires me to | |
| 4993 * duplicate the OpenAL queued buffers state with my own | |
| 4994 * code and try to derive what the current playing buffer is by indirect observation of | |
| 4995 * looking at buffers_processed. But of course this has a ton of downsides since my | |
| 4996 * queries do not give me perfect timing of what OpenAL is actually doing and | |
| 4997 * the fact that some of the distributions seem to have buffer queuing problems | |
| 4998 * with their query results (CoreAudio). This also means a ton of extra code | |
| 4999 * on my side. The lack of support of a 1 line call has required me to | |
| 5000 * implement yet another entire state machine. <sigh> | |
| 5001 */ | |
| 5002 #if 0 /* This code will not work until possibly OpenAL 1.1 because of Nvidia */ | |
| 5003 /* Get the id to the current buffer playing */ | |
| 5004 alGetSourcei( | |
| 5005 ALmixer_Channel_List[i].alsource, | |
| 5006 AL_BUFFER, ¤t_buffer_id | |
| 5007 ); | |
| 5008 if((error = alGetError()) != AL_NO_ERROR) | |
| 5009 { | |
| 5010 fprintf(stderr, "53Testing error: %s\n", | |
|
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5011 alGetString(error)); |
| 0 | 5012 } |
| 5013 | |
| 5014 /* Before the hard stuff, check to see if the | |
| 5015 * current queued AL buffer has changed. | |
| 5016 * If it has, we should launch a data callback if | |
| 5017 * necessary | |
| 5018 */ | |
| 5019 if( ((ALuint)current_buffer_id) != | |
| 5020 ALmixer_Channel_List[i].almixer_data->current_buffer) | |
| 5021 { | |
| 5022 ALmixer_Channel_List[i].almixer_data->current_buffer | |
| 5023 = (ALuint)current_buffer_id; | |
| 5024 | |
| 5025 Invoke_Streamed_Channel_Data_Callback(i, ALmixer_Channel_List[i].almixer_data, current_buffer_id); | |
| 5026 } | |
| 5027 #else | |
| 5028 /* Only do this if "access_data" was requested (i.e. the circular_buffer!=NULL) | |
| 5029 * And if one of the two are true: | |
| 5030 * Either buffers_processed > 0 (because the current_buffer might have changed) | |
| 5031 * or if the current_buffer==0 (because we are in an initial state or recovering from | |
| 5032 * a buffer underrun) | |
| 5033 */ | |
| 5034 if((ALmixer_Channel_List[i].almixer_data->circular_buffer_queue != NULL) | |
| 5035 && ( | |
| 5036 (buffers_processed > 0) || (0 == ALmixer_Channel_List[i].almixer_data->current_buffer) | |
| 5037 ) | |
| 5038 ) | |
| 5039 { | |
|
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5040 ALint k; |
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5041 ALuint queue_ret_flag; |
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5042 ALubyte is_out_of_sync = 0; |
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5043 ALuint my_queue_size = CircularQueueUnsignedInt_Size(ALmixer_Channel_List[i].almixer_data->circular_buffer_queue); |
| 1 | 5044 /* Ugh, I have to deal with signed/unsigned mismatch here. */ |
| 5045 ALint buffers_unplayed_int = buffers_still_queued - buffers_processed; | |
|
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5046 ALuint unplayed_buffers; |
| 1 | 5047 if(buffers_unplayed_int < 0) |
| 5048 { | |
| 5049 unplayed_buffers = 0; | |
| 5050 } | |
| 5051 else | |
| 5052 { | |
|
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5053 unplayed_buffers = (ALuint)buffers_unplayed_int; |
| 1 | 5054 } |
| 0 | 5055 /* |
| 5056 fprintf(stderr, "Queue in processed check, before pop, buffers_processed=%d\n", buffers_processed); | |
| 5057 CircularQueueUnsignedInt_Print(ALmixer_Channel_List[i].almixer_data->circular_buffer_queue); | |
| 5058 */ | |
| 5059 /* We can't make any determinations solely based on the number of buffers_processed | |
| 5060 * because currently, we only unqueue 1 buffer per loop. That means if 2 or more | |
| 5061 * buffers became processed in one loop, the following loop, we would have | |
| 5062 * at least that_many-1 buffers_processed (plus possible new processed). | |
| 5063 * If we tried to just remove 1 buffer from our queue, we would be incorrect | |
| 5064 * because we would not actually reflect the current playing buffer. | |
| 5065 * So the solution seems to be to make sure our queue is the same size | |
| 5066 * as the number of buffers_queued-buffers_processed, and return the head of our queue | |
| 5067 * as the current playing buffer. | |
| 5068 */ | |
| 5069 /* Also, we have a corner case. When we first start playing or if we have | |
| 5070 * a buffer underrun, we have not done a data callback. | |
| 5071 * In this case, we need to see if there is any new data in our queue | |
| 5072 * and if so, launch that data callback. | |
| 5073 */ | |
| 5074 /* Warning, this code risks the possibility of no data callback being fired if | |
| 5075 * the system is really late (or skipped buffers). | |
| 5076 */ | |
| 5077 | |
| 5078 /* First, let's syncronize our queue with the OpenAL queue */ | |
| 5079 #if 0 | |
| 5080 fprintf(stderr, "inside, Buffers processed=%d, Buffers queued=%d, my queue=%d\n", | |
| 5081 buffers_processed, buffers_still_queued, my_queue_size); | |
| 1 | 5082 #endif |
|
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5083 is_out_of_sync = 1; |
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5084 for(k=0; k<buffers_processed; k++) |
| 0 | 5085 { |
|
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5086 queue_ret_flag = CircularQueueUnsignedInt_PopFront( |
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|
5087 ALmixer_Channel_List[i].almixer_data->circular_buffer_queue); |
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5088 if(0 == queue_ret_flag) |
| 0 | 5089 { |
|
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5090 fprintf(stderr, "53 Error popping queue\n"); |
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5091 } |
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5092 } |
| 0 | 5093 my_queue_size = CircularQueueUnsignedInt_Size(ALmixer_Channel_List[i].almixer_data->circular_buffer_queue); |
| 5094 /* We have several possibilities we need to handle: | |
| 5095 * 1) We are in an initial state or underrun and need to do a data callback on the head. | |
| 5096 * 2) We were out of sync and need to do a new data callback on the new head. | |
| 5097 * 3) We were not out of sync but just had left over processed buffers which caused us to | |
| 5098 * fall in this block of code. (Don't do anything.) | |
| 5099 */ | |
| 5100 if( (0 == ALmixer_Channel_List[i].almixer_data->current_buffer) || (1 == is_out_of_sync) ) | |
| 5101 { | |
| 5102 if(my_queue_size > 0) | |
| 5103 { | |
| 5104 current_buffer_id = CircularQueueUnsignedInt_Front( | |
| 5105 ALmixer_Channel_List[i].almixer_data->circular_buffer_queue); | |
| 5106 if(0 == current_buffer_id) | |
| 5107 { | |
| 5108 fprintf(stderr, "53a Internal Error, current_buffer_id=0 when it shouldn't be 0\n"); | |
| 5109 } | |
| 5110 /* | |
| 5111 else | |
| 5112 { | |
| 5113 fprintf(stderr, "Queue in processed check, after pop\n"); | |
| 5114 CircularQueueUnsignedInt_Print(ALmixer_Channel_List[i].almixer_data->circular_buffer_queue); | |
| 5115 } | |
| 5116 */ | |
| 5117 ALmixer_Channel_List[i].almixer_data->current_buffer | |
| 5118 = (ALuint)current_buffer_id; | |
| 5119 | |
| 5120 #if 0 | |
| 5121 /* Remove me...only for checking...doesn't work on Nvidia */ | |
| 5122 { | |
| 5123 ALuint real_id; | |
| 5124 alGetSourcei( | |
| 5125 ALmixer_Channel_List[i].alsource, | |
| 5126 AL_BUFFER, &real_id | |
| 5127 ); | |
| 5128 alGetError(); | |
| 5129 fprintf(stderr, "Callback fired on data buffer=%d, real_id shoud be=%d\n", current_buffer_id, real_id); | |
| 5130 } | |
| 5131 #endif | |
| 5132 Invoke_Streamed_Channel_Data_Callback(i, ALmixer_Channel_List[i].almixer_data, current_buffer_id); | |
| 5133 } | |
| 5134 else | |
| 5135 { | |
| 1 | 5136 /* |
| 0 | 5137 fprintf(stderr, "53b, Notice/Warning:, OpenAL queue has been depleted.\n"); |
| 1 | 5138 PrintQueueStatus(ALmixer_Channel_List[i].alsource); |
| 5139 */ | |
| 0 | 5140 /* In this case, we might either be in an underrun or finished with playback */ |
| 5141 ALmixer_Channel_List[i].almixer_data->current_buffer = 0; | |
| 5142 } | |
| 5143 } | |
| 5144 } | |
| 5145 #endif | |
| 5146 | |
| 5147 | |
| 5148 | |
| 5149 /* Just a test - remove | |
| 5150 if( ALmixer_Channel_List[i].loops > 0) | |
| 5151 { | |
| 5152 fprintf(stderr, ">>>>>>>>>>>>>>>Loops = %d\n", | |
| 5153 ALmixer_Channel_List[i].loops); | |
| 5154 } | |
| 5155 */ | |
| 5156 #if 0 | |
| 5157 fprintf(stderr, "Buffers processed = %d\n", buffers_processed); | |
| 5158 fprintf(stderr, "Buffers queued= %d\n", buffers_still_queued); | |
| 5159 #endif | |
| 5160 /* We've used up a buffer so we need to unqueue and replace */ | |
| 5161 /* Okay, it gets more complicated here: | |
| 5162 * We need to Queue more data | |
| 5163 * if buffers_processed > 0 or | |
| 5164 * if num_of_buffers_in_use < NUMBER_OF_QUEUE_BUFFERS | |
| 5165 * but we don't do this if at EOF, | |
| 5166 * except when there is looping | |
| 5167 */ | |
| 5168 /* For this to work, we must rely on EVERYTHING | |
| 5169 * else to unset the EOF if there is looping. | |
| 5170 * Remember, even Play() must do this | |
| 5171 */ | |
| 5172 | |
| 5173 /* If not EOF, then we are still playing. | |
| 5174 * Inside, we might find num_of_buffers < NUM...QUEUE_BUF.. | |
| 5175 * or buffers_process > 0 | |
| 5176 * in which case we queue up. | |
| 5177 * We also might find no buffers we need to fill, | |
| 5178 * in which case we just keep going | |
| 5179 */ | |
| 5180 if( ! ALmixer_Channel_List[i].almixer_data->eof) | |
| 5181 { | |
|
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5182 ALuint bytes_returned; |
| 0 | 5183 /* We have a priority. We first must assign |
| 5184 * unused buffers in reserve. If there is nothing | |
| 5185 * left, then we may unqueue buffers. We can't | |
| 5186 * do it the other way around because we will | |
| 5187 * lose the pointer to the unqueued buffer | |
| 5188 */ | |
| 5189 if(ALmixer_Channel_List[i].almixer_data->num_buffers_in_use | |
| 5190 < ALmixer_Channel_List[i].almixer_data->max_queue_buffers) | |
| 5191 { | |
| 1 | 5192 #if 0 |
| 0 | 5193 fprintf(stderr, "Getting more data in NOT_EOF and num_buffers_in_use (%d) < max_queue (%d)\n", |
| 5194 ALmixer_Channel_List[i].almixer_data->num_buffers_in_use, | |
| 5195 ALmixer_Channel_List[i].almixer_data->max_queue_buffers); | |
| 1 | 5196 #endif |
| 0 | 5197 /* Going to add an unused packet. |
| 5198 * Grab next packet */ | |
| 5199 bytes_returned = GetMoreData( | |
| 5200 ALmixer_Channel_List[i].almixer_data, | |
| 5201 ALmixer_Channel_List[i].almixer_data->buffer[ | |
| 5202 ALmixer_Channel_List[i].almixer_data->num_buffers_in_use] | |
| 5203 ); | |
| 5204 } | |
| 5205 /* For processed > 0 */ | |
| 5206 else if(buffers_processed > 0) | |
| 5207 { | |
| 5208 /* Unqueue only 1 buffer for now. | |
| 5209 * If there are more than one, | |
| 5210 * let the next Update pass deal with it | |
| 5211 * so we don't stall the program for too long. | |
| 5212 */ | |
| 5213 #if 0 | |
| 5214 fprintf(stderr, "About to Unqueue, Buffers processed = %d\n", buffers_processed); | |
| 5215 fprintf(stderr, "Buffers queued= %d\n", buffers_still_queued); | |
| 5216 fprintf(stderr, "Unqueuing a buffer\n"); | |
| 5217 #endif | |
| 5218 alSourceUnqueueBuffers( | |
| 5219 ALmixer_Channel_List[i].alsource, | |
| 5220 1, &unqueued_buffer_id | |
| 5221 ); | |
| 5222 if((error = alGetError()) != AL_NO_ERROR) | |
| 5223 { | |
| 5224 fprintf(stderr, "Error with unqueue: %s", | |
|
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5225 alGetString(error)); |
| 0 | 5226 ALmixer_SetError("Unqueue buffer failed: %s", |
|
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5227 alGetString(error) ); |
| 0 | 5228 error_flag--; |
| 5229 } | |
| 5230 /* | |
| 5231 fprintf(stderr, "Right after unqueue..."); | |
| 5232 PrintQueueStatus(ALmixer_Channel_List[i].alsource); | |
| 5233 fprintf(stderr, "Getting more data for NOT_EOF, max_buffers filled\n"); | |
| 5234 */ | |
| 5235 /* Grab unqueued packet */ | |
| 5236 bytes_returned = GetMoreData( | |
| 5237 ALmixer_Channel_List[i].almixer_data, | |
| 5238 unqueued_buffer_id); | |
| 5239 } | |
| 5240 /* We are still streaming, but currently | |
| 5241 * don't need to fill any buffers */ | |
| 5242 else | |
| 5243 { | |
| 5244 /* Might want to check state */ | |
| 5245 /* In case the playback stopped, | |
|
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5246 * we need to resume |
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5247 * a.k.a. buffer underrun |
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5248 */ |
| 1 | 5249 #if 1 |
| 5250 /* Try not refetching the state here because I'm getting a duplicate | |
| 5251 buffer playback (hiccup) */ | |
| 0 | 5252 alGetSourcei( |
| 5253 ALmixer_Channel_List[i].alsource, | |
| 5254 AL_SOURCE_STATE, &state | |
| 5255 ); | |
| 1 | 5256 if((error = alGetError()) != AL_NO_ERROR) |
| 5257 { | |
| 5258 fprintf(stderr, "54bTesting error: %s\n", | |
|
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5259 alGetString(error)); |
| 1 | 5260 } |
| 5261 /* Get the number of buffers processed | |
| 5262 */ | |
| 5263 alGetSourcei( | |
| 5264 ALmixer_Channel_List[i].alsource, | |
| 5265 AL_BUFFERS_PROCESSED, | |
| 5266 &buffers_processed | |
| 5267 ); | |
| 5268 if((error = alGetError()) != AL_NO_ERROR) | |
| 5269 { | |
| 5270 fprintf(stderr, "54cError, Can't get buffers_processed: %s\n", | |
|
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5271 alGetString(error)); |
| 1 | 5272 } |
| 5273 #endif | |
| 0 | 5274 if(AL_STOPPED == state) |
| 5275 { | |
| 5276 /* Resuming in not eof, but nothing to buffer */ | |
| 1 | 5277 |
| 5278 /* Okay, here's another lately discovered problem: | |
| 5279 * I can't find it in the spec, but for at least some of the | |
| 5280 * implementations, if I call play on a stopped source that | |
| 5281 * has processed buffers, all those buffers get marked as unprocessed | |
| 5282 * on alSourcePlay. So if I had a queue of 25 with 24 of the buffers | |
| 5283 * processed, on resume, the earlier 24 buffers will get replayed, | |
| 5284 * causing a "hiccup" like sound in the playback. | |
| 5285 * To avoid this, I must unqueue all processed buffers before | |
| 5286 * calling play. But to complicate things, I need to worry about resyncing | |
| 5287 * the circular queue with this since I designed this thing | |
| 5288 * with some correlation between the two. However, I might | |
| 5289 * have already handled this, so I will try writing this code without | |
| 5290 * syncing for now. | |
| 5291 * There is currently an assumption that a buffer | |
| 5292 * was queued above so I actually have something | |
| 5293 * to play. | |
| 5294 */ | |
| 5295 ALint temp_count; | |
|
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5296 #if 0 |
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5297 fprintf(stderr, "STOPPED1, need to clear processed=%d, status is:\n", buffers_processed); |
| 1 | 5298 PrintQueueStatus(ALmixer_Channel_List[i].alsource); |
|
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5299 #endif |
| 1 | 5300 for(temp_count=0; temp_count<buffers_processed; temp_count++) |
| 5301 { | |
| 5302 alSourceUnqueueBuffers( | |
| 5303 ALmixer_Channel_List[i].alsource, | |
| 5304 1, &unqueued_buffer_id | |
| 5305 ); | |
| 5306 if((error = alGetError()) != AL_NO_ERROR) | |
| 5307 { | |
| 5308 fprintf(stderr, "55aTesting error: %s\n", | |
|
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5309 alGetString(error)); |
| 1 | 5310 error_flag--; |
| 5311 } | |
| 5312 } | |
|
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5313 #if 0 |
| 1 | 5314 fprintf(stderr, "After unqueue clear...:\n"); |
| 5315 PrintQueueStatus(ALmixer_Channel_List[i].alsource); | |
|
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5316 #endif |
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5317 /* My assertion: We are STOPPED but not EOF. |
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5318 * This means we have a buffer underrun. |
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5319 * We just cleared out the unqueued buffers. |
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5320 * So we need to reset the mixer_data to reflect we have |
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5321 * no buffers in queue. |
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5322 * We need to GetMoreData and then queue up the data. |
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5323 * Then we need to resume playing. |
|
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|
5324 */ |
|
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|
5325 #if 0 |
|
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|
5326 int buffers_queued; |
|
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|
5327 alGetSourcei( |
|
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|
5328 ALmixer_Channel_List[i].alsource, |
|
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|
5329 AL_BUFFERS_QUEUED, |
|
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|
5330 &buffers_queued |
|
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5331 ); |
|
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|
5332 |
| 1 | 5333 if((error = alGetError()) != AL_NO_ERROR) |
| 5334 { | |
|
2
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5335 fprintf(stderr, "Error in PrintQueueStatus, Can't get buffers_queued: %s\n", |
|
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|
5336 alGetString(error)); |
|
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|
5337 } |
|
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|
5338 assert(buffers_queued == 0); |
|
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5339 fprintf(stderr, "buffer underrun: buffers_queued:%d\n", buffers_queued); |
|
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5340 #endif |
|
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|
5341 |
|
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|
5342 /* Reset the number of buffers in use to 0 */ |
|
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|
5343 ALmixer_Channel_List[i].almixer_data->num_buffers_in_use = 0; |
|
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|
5344 |
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|
5345 /* Get more data and put it in the first buffer */ |
|
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|
5346 bytes_returned = GetMoreData( |
|
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|
5347 ALmixer_Channel_List[i].almixer_data, |
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|
5348 ALmixer_Channel_List[i].almixer_data->buffer[0] |
|
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5349 ); |
|
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|
5350 /* NOTE: We might want to look for EOF and handle it here. |
|
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5351 * Currently, I just let the next loop handle it which seems to be working. |
|
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5352 */ |
|
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|
5353 if(bytes_returned > 0) |
|
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5354 { |
|
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|
5355 /* Queue up the new data */ |
|
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|
5356 alSourceQueueBuffers( |
|
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|
5357 ALmixer_Channel_List[i].alsource, |
|
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5358 1, |
|
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|
5359 &ALmixer_Channel_List[i].almixer_data->buffer[0] |
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|
5360 ); |
|
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|
5361 if((error = alGetError()) != AL_NO_ERROR) |
|
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5362 { |
|
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|
5363 fprintf(stderr, "56e alSourceQueueBuffers error: %s\n", |
|
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|
5364 alGetString(error)); |
|
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|
5365 } |
|
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|
5366 /* Increment the number of buffers in use */ |
|
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|
5367 ALmixer_Channel_List[i].almixer_data->num_buffers_in_use++; |
|
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|
5368 |
|
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5369 |
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|
5370 /* We need to empty and update the circular buffer queue if it is in use */ |
|
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|
5371 if(ALmixer_Channel_List[i].almixer_data->circular_buffer_queue != NULL) |
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5372 { |
|
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|
5373 ALuint queue_ret_flag; |
|
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|
5374 CircularQueueUnsignedInt_Clear(ALmixer_Channel_List[i].almixer_data->circular_buffer_queue); |
|
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|
5375 queue_ret_flag = CircularQueueUnsignedInt_PushBack( |
|
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|
5376 ALmixer_Channel_List[i].almixer_data->circular_buffer_queue, |
|
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|
5377 ALmixer_Channel_List[i].almixer_data->buffer[0] |
|
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|
5378 ); |
|
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|
5379 if(0 == queue_ret_flag) |
|
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|
5380 { |
|
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|
5381 fprintf(stderr, "56fSerious internal error: CircularQueue could not push into queue.\n"); |
|
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|
5382 ALmixer_SetError("Serious internal error: CircularQueue failed to push into queue"); |
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5383 } |
|
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5384 } |
|
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5385 |
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5386 |
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5387 |
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5388 |
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5389 /* Resume playback from underrun */ |
|
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|
5390 alSourcePlay(ALmixer_Channel_List[i].alsource); |
|
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|
5391 if((error = alGetError()) != AL_NO_ERROR) |
|
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|
5392 { |
|
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|
5393 fprintf(stderr, "55Tbesting error: %s\n", |
|
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|
5394 alGetString(error)); |
|
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|
5395 } |
| 1 | 5396 } |
|
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5397 |
| 0 | 5398 } |
| 5399 /* Let's escape to the next loop. | |
| 5400 * All code below this point is for queuing up | |
| 5401 */ | |
| 5402 /* | |
| 1 | 5403 fprintf(stderr, "Entry: Nothing to do...continue\n\n"); |
| 5404 */ | |
| 0 | 5405 continue; |
| 5406 } | |
| 5407 /* We now know we have to fill an available | |
| 5408 * buffer. | |
| 5409 */ | |
| 5410 | |
| 5411 /* In the previous branch, we just grabbed more data. | |
| 5412 * Let's check it to make sure it's okay, | |
| 5413 * and then queue it up | |
| 5414 */ | |
|
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5415 /* This check doesn't work anymore because it is now ALuint */ |
| 0 | 5416 #if 0 |
| 5417 if(-1 == bytes_returned) | |
| 5418 { | |
| 5419 /* Problem occurred...not sure what to do */ | |
| 5420 /* Go to next loop? */ | |
| 5421 error_flag--; | |
| 5422 /* Set the eof flag to force a quit so | |
| 5423 * we don't get stuck in an infinite loop | |
| 5424 */ | |
| 5425 ALmixer_Channel_List[i].almixer_data->eof = 1; | |
| 5426 continue; | |
| 5427 } | |
| 5428 #endif | |
| 5429 /* This is a special case where we've run | |
| 5430 * out of data. We should check for loops | |
| 5431 * and get more data. If there is no loop, | |
| 5432 * then do nothing and wait for future | |
| 5433 * update passes to handle the EOF. | |
| 5434 * The advantage of handling the loop here | |
| 5435 * instead of waiting for play to stop is | |
| 5436 * that we should be able to keep the buffer | |
| 5437 * filled. | |
| 5438 */ | |
| 5439 #if 0 | |
| 5440 else if(0 == bytes_returned) | |
| 5441 #endif | |
| 5442 if(0 == bytes_returned) | |
| 5443 { | |
|
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5444 /* |
| 0 | 5445 fprintf(stderr, "We got 0 bytes from reading. Checking for loops\n"); |
|
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5446 */ |
| 0 | 5447 /* Check for loops */ |
| 5448 if( ALmixer_Channel_List[i].loops != 0 ) | |
| 5449 { | |
| 5450 /* We have to loop, so rewind | |
| 5451 * and fetch more data | |
| 5452 */ | |
|
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5453 /* |
| 0 | 5454 fprintf(stderr, "Rewinding data\n"); |
|
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|
5455 */ |
| 0 | 5456 if(0 == Sound_Rewind( |
| 5457 ALmixer_Channel_List[i].almixer_data->sample)) | |
| 5458 { | |
| 5459 fprintf(stderr, "Rewinding failed\n"); | |
| 5460 ALmixer_SetError( Sound_GetError() ); | |
| 5461 ALmixer_Channel_List[i].loops = 0; | |
| 5462 error_flag--; | |
| 5463 /* We'll continue on because we do have some valid data */ | |
| 5464 continue; | |
| 5465 } | |
| 5466 /* Remember to reset the data->eof flag */ | |
| 5467 ALmixer_Channel_List[i].almixer_data->eof = 0; | |
| 5468 if(ALmixer_Channel_List[i].loops > 0) | |
| 5469 { | |
| 5470 ALmixer_Channel_List[i].loops--; | |
| 5471 } | |
| 5472 /* Try grabbing another packet now. | |
| 5473 * Since we may have already unqueued a | |
| 5474 * buffer, we don't want to lose it. | |
| 5475 */ | |
| 5476 if(ALmixer_Channel_List[i].almixer_data->num_buffers_in_use | |
| 5477 < ALmixer_Channel_List[i].almixer_data->max_queue_buffers) | |
| 5478 { | |
|
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5479 /* |
| 0 | 5480 fprintf(stderr, "We got %d bytes from reading loop. Filling unused packet\n", bytes_returned); |
|
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5481 */ |
| 0 | 5482 /* Grab next packet */ |
| 5483 bytes_returned = GetMoreData( | |
| 5484 ALmixer_Channel_List[i].almixer_data, | |
| 5485 ALmixer_Channel_List[i].almixer_data->buffer[ | |
| 5486 ALmixer_Channel_List[i].almixer_data->num_buffers_in_use] | |
| 5487 ); | |
|
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5488 /* |
| 0 | 5489 fprintf(stderr, "We reread %d bytes into unused packet\n", bytes_returned); |
|
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5490 */ |
| 0 | 5491 } |
| 5492 /* Refilling unqueued packet */ | |
| 5493 else | |
| 5494 { | |
|
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5495 /* |
| 0 | 5496 fprintf(stderr, "We got %d bytes from reading loop. Filling unqueued packet\n", bytes_returned); |
|
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5497 */ |
| 0 | 5498 /* Grab next packet */ |
| 5499 bytes_returned = GetMoreData( | |
| 5500 ALmixer_Channel_List[i].almixer_data, | |
| 5501 unqueued_buffer_id); | |
|
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5502 /* |
| 0 | 5503 fprintf(stderr, "We reread %d bytes into unqueued packet\n", bytes_returned); |
|
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|
5504 */ |
| 0 | 5505 } |
| 5506 /* Another error check */ | |
| 5507 /* | |
| 5508 if(bytes_returned <= 0) | |
| 5509 */ | |
| 5510 if(0 == bytes_returned) | |
| 5511 { | |
| 5512 fprintf(stderr, "??????????ERROR\n"); | |
| 5513 ALmixer_SetError("Could not loop because after rewind, no data could be retrieved"); | |
| 5514 /* Problem occurred...not sure what to do */ | |
| 5515 /* Go to next loop? */ | |
| 5516 error_flag--; | |
| 5517 /* Set the eof flag to force a quit so | |
| 5518 * we don't get stuck in an infinite loop | |
| 5519 */ | |
| 5520 ALmixer_Channel_List[i].almixer_data->eof = 1; | |
| 5521 continue; | |
| 5522 } | |
| 5523 /* We made it to the end. We still need | |
| 5524 * to BufferData, so let this branch | |
| 5525 * fall into the next piece of | |
| 5526 * code below which will handle that | |
| 5527 */ | |
| 5528 | |
| 5529 | |
| 5530 } /* END loop check */ | |
| 5531 else | |
| 5532 { | |
| 5533 /* No more loops to do. | |
| 5534 * EOF flag should be set. | |
| 5535 * Just go to next loop and | |
| 5536 * let things be handled correctly | |
| 5537 * in future update calls | |
| 5538 */ | |
| 1 | 5539 /* |
| 0 | 5540 fprintf(stderr, "SHOULD BE EOF\n"); |
| 5541 | |
| 5542 PrintQueueStatus(ALmixer_Channel_List[i].alsource); | |
| 1 | 5543 */ |
| 0 | 5544 continue; |
| 5545 } | |
| 5546 } /* END if bytes_returned == 0 */ | |
| 5547 /********* Possible trouble point. I might be queueing empty buffers on the mac. | |
| 5548 * This check doesn't say if the buffer is valid. Only the EOF assumption is a clue at this point | |
| 5549 */ | |
| 5550 /* Fall here */ | |
| 5551 /* Everything is normal. We aren't | |
| 5552 * at an EOF, but need to simply | |
| 5553 * queue more data. The data is already checked for good, | |
| 5554 * so queue it up */ | |
| 5555 if(ALmixer_Channel_List[i].almixer_data->num_buffers_in_use | |
| 5556 < ALmixer_Channel_List[i].almixer_data->max_queue_buffers) | |
| 5557 { | |
| 5558 /* Keep count of how many buffers we have | |
| 5559 * to queue so we can return the value | |
| 5560 */ | |
| 5561 retval++; | |
| 1 | 5562 /* |
| 0 | 5563 fprintf(stderr, "NOT_EOF???, about to Queue more data for num_buffers (%d) < max_queue (%d)\n", |
| 5564 ALmixer_Channel_List[i].almixer_data->num_buffers_in_use, | |
| 5565 ALmixer_Channel_List[i].almixer_data->max_queue_buffers); | |
| 1 | 5566 */ |
| 0 | 5567 alSourceQueueBuffers( |
| 5568 ALmixer_Channel_List[i].alsource, | |
| 5569 1, | |
| 5570 &ALmixer_Channel_List[i].almixer_data->buffer[ | |
| 5571 ALmixer_Channel_List[i].almixer_data->num_buffers_in_use] | |
| 5572 ); | |
| 5573 if((error = alGetError()) != AL_NO_ERROR) | |
| 5574 { | |
| 5575 fprintf(stderr, "56Testing error: %s\n", | |
|
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diff
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|
5576 alGetString(error)); |
| 0 | 5577 } |
| 5578 /* This is part of the hideous Nvidia workaround. In order to figure out | |
| 5579 * which buffer to show during callbacks (for things like | |
| 5580 * o-scopes), I must keep a copy of the buffers that are queued in my own | |
| 5581 * data structure. This code will be called only if | |
| 5582 * "access_data" was set, indicated by whether the queue is NULL. | |
| 5583 */ | |
| 5584 if(ALmixer_Channel_List[i].almixer_data->circular_buffer_queue != NULL) | |
| 5585 { | |
|
2
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diff
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|
5586 ALuint queue_ret_flag; |
|
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parents:
1
diff
changeset
|
5587 // fprintf(stderr, "56d: CircularQueue_PushBack.\n"); |
| 0 | 5588 queue_ret_flag = CircularQueueUnsignedInt_PushBack( |
| 5589 ALmixer_Channel_List[i].almixer_data->circular_buffer_queue, | |
| 5590 ALmixer_Channel_List[i].almixer_data->buffer[ALmixer_Channel_List[i].almixer_data->num_buffers_in_use] | |
| 5591 ); | |
| 5592 if(0 == queue_ret_flag) | |
| 5593 { | |
| 5594 fprintf(stderr, "56aSerious internal error: CircularQueue could not push into queue.\n"); | |
| 5595 ALmixer_SetError("Serious internal error: CircularQueue failed to push into queue"); | |
| 5596 } | |
| 5597 /* | |
| 5598 else | |
| 5599 { | |
| 5600 CircularQueueUnsignedInt_Print(ALmixer_Channel_List[i].almixer_data->circular_buffer_queue); | |
| 5601 } | |
| 5602 */ | |
| 5603 } | |
| 5604 } | |
| 5605 /* for processed > 0 */ | |
| 5606 else | |
| 5607 { | |
| 5608 /* Keep count of how many buffers we have | |
| 5609 * to queue so we can return the value | |
| 5610 */ | |
| 5611 retval++; | |
| 5612 /* | |
| 5613 fprintf(stderr, "NOT_EOF, about to Queue more data for filled max_queue (%d)\n", | |
| 5614 ALmixer_Channel_List[i].almixer_data->max_queue_buffers); | |
| 5615 */ | |
| 5616 alSourceQueueBuffers( | |
| 5617 ALmixer_Channel_List[i].alsource, | |
| 5618 1, &unqueued_buffer_id); | |
| 5619 if((error = alGetError()) != AL_NO_ERROR) | |
| 5620 { | |
| 5621 ALmixer_SetError("Could not QueueBuffer: %s", | |
|
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1
diff
changeset
|
5622 alGetString(error) ); |
| 0 | 5623 error_flag--; |
| 5624 continue; | |
| 5625 } | |
| 5626 /* This is part of the hideous Nvidia workaround. In order to figure out | |
| 5627 * which buffer to show during callbacks (for things like | |
| 5628 * o-scopes), I must keep a copy of the buffers that are queued in my own | |
| 5629 * data structure. This code will be called only if | |
| 5630 * "access_data" was set, indicated by whether the queue is NULL. | |
| 5631 */ | |
| 5632 if(ALmixer_Channel_List[i].almixer_data->circular_buffer_queue != NULL) | |
| 5633 { | |
|
2
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diff
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|
5634 ALuint queue_ret_flag; |
|
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Overhaul prep for first public release.
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parents:
1
diff
changeset
|
5635 // fprintf(stderr, "56e: CircularQueue_PushBack.\n"); |
| 0 | 5636 queue_ret_flag = CircularQueueUnsignedInt_PushBack( |
| 5637 ALmixer_Channel_List[i].almixer_data->circular_buffer_queue, | |
| 5638 unqueued_buffer_id | |
| 5639 ); | |
| 5640 if(0 == queue_ret_flag) | |
| 5641 { | |
| 5642 fprintf(stderr, "56bSerious internal error: CircularQueue could not push into queue.\n"); | |
| 5643 ALmixer_SetError("Serious internal error: CircularQueue failed to push into queue"); | |
| 5644 } | |
|
2
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Eric Wing <ewing . public |-at-| gmail . com>
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1
diff
changeset
|
5645 #if 0 |
| 0 | 5646 else |
| 5647 { | |
| 5648 CircularQueueUnsignedInt_Print(ALmixer_Channel_List[i].almixer_data->circular_buffer_queue); | |
| 5649 } | |
|
2
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diff
changeset
|
5650 #endif |
| 0 | 5651 } |
| 5652 } | |
| 5653 /* If we used an available buffer queue, | |
| 5654 * then we need to update the number of them in use | |
| 5655 */ | |
| 5656 if(ALmixer_Channel_List[i].almixer_data->num_buffers_in_use | |
| 5657 < ALmixer_Channel_List[i].almixer_data->max_queue_buffers) | |
| 5658 { | |
| 5659 /* Increment the number of buffers in use */ | |
| 5660 ALmixer_Channel_List[i].almixer_data->num_buffers_in_use++; | |
| 5661 } | |
| 5662 /* Might want to check state */ | |
| 5663 /* In case the playback stopped, | |
| 5664 * we need to resume */ | |
| 1 | 5665 #if 1 |
| 5666 /* Try not refetching the state here because I'm getting a duplicate | |
| 5667 buffer playback (hiccup) */ | |
| 0 | 5668 alGetSourcei( |
| 5669 ALmixer_Channel_List[i].alsource, | |
| 5670 AL_SOURCE_STATE, &state | |
| 5671 ); | |
| 1 | 5672 if((error = alGetError()) != AL_NO_ERROR) |
| 5673 { | |
| 5674 fprintf(stderr, "57bTesting error: %s\n", | |
|
2
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1
diff
changeset
|
5675 alGetString(error)); |
| 1 | 5676 } |
| 5677 /* Get the number of buffers processed | |
| 5678 */ | |
| 5679 alGetSourcei( | |
| 5680 ALmixer_Channel_List[i].alsource, | |
| 5681 AL_BUFFERS_PROCESSED, | |
| 5682 &buffers_processed | |
| 5683 ); | |
| 5684 if((error = alGetError()) != AL_NO_ERROR) | |
| 5685 { | |
| 5686 fprintf(stderr, "57cError, Can't get buffers_processed: %s\n", | |
|
2
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1
diff
changeset
|
5687 alGetString(error)); |
| 1 | 5688 } |
| 5689 #endif | |
| 0 | 5690 if(AL_STOPPED == state) |
| 5691 { | |
| 1 | 5692 /* |
| 0 | 5693 fprintf(stderr, "Resuming in not eof\n"); |
| 1 | 5694 */ |
| 5695 /* Okay, here's another lately discovered problem: | |
| 5696 * I can't find it in the spec, but for at least some of the | |
| 5697 * implementations, if I call play on a stopped source that | |
| 5698 * has processed buffers, all those buffers get marked as unprocessed | |
| 5699 * on alSourcePlay. So if I had a queue of 25 with 24 of the buffers | |
| 5700 * processed, on resume, the earlier 24 buffers will get replayed, | |
| 5701 * causing a "hiccup" like sound in the playback. | |
| 5702 * To avoid this, I must unqueue all processed buffers before | |
| 5703 * calling play. But to complicate things, I need to worry about resyncing | |
| 5704 * the circular queue with this since I designed this thing | |
| 5705 * with some correlation between the two. However, I might | |
| 5706 * have already handled this, so I will try writing this code without | |
| 5707 * syncing for now. | |
| 5708 * There is currently an assumption that a buffer | |
| 5709 * was queued above so I actually have something | |
| 5710 * to play. | |
| 5711 */ | |
| 5712 ALint temp_count; | |
| 5713 /* | |
| 5714 fprintf(stderr, "STOPPED2, need to clear processed, status is:\n"); | |
| 5715 PrintQueueStatus(ALmixer_Channel_List[i].alsource); | |
| 5716 */ | |
| 5717 | |
| 5718 for(temp_count=0; temp_count<buffers_processed; temp_count++) | |
| 5719 { | |
| 5720 alSourceUnqueueBuffers( | |
| 5721 ALmixer_Channel_List[i].alsource, | |
| 5722 1, &unqueued_buffer_id | |
| 5723 ); | |
| 5724 if((error = alGetError()) != AL_NO_ERROR) | |
| 5725 { | |
| 5726 fprintf(stderr, "58aTesting error: %s\n", | |
|
2
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1
diff
changeset
|
5727 alGetString(error)); |
| 1 | 5728 error_flag--; |
| 5729 } | |
| 5730 } | |
| 5731 /* | |
| 5732 fprintf(stderr, "After unqueue clear...:\n"); | |
| 5733 PrintQueueStatus(ALmixer_Channel_List[i].alsource); | |
| 5734 */ | |
| 5735 | |
| 5736 alSourcePlay(ALmixer_Channel_List[i].alsource); | |
| 5737 if((error = alGetError()) != AL_NO_ERROR) | |
| 5738 { | |
| 5739 fprintf(stderr, "55Tbesting 8rror: %s\n", | |
|
2
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Eric Wing <ewing . public |-at-| gmail . com>
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1
diff
changeset
|
5740 alGetString(error)); |
| 1 | 5741 } |
| 0 | 5742 } |
| 5743 continue; | |
| 5744 } /* END if( ! eof) */ | |
| 5745 /* We have hit EOF in the SDL_Sound sample and there | |
| 5746 * are no more loops. However, there may still be | |
| 5747 * buffers in the OpenAL queue which still need to | |
| 5748 * be played out. The following body of code will | |
| 5749 * determine if play is still happening or | |
| 5750 * initiate the stop/cleanup sequenece. | |
| 5751 */ | |
| 5752 else | |
| 5753 { | |
| 5754 /* Let's continue to remove the used up | |
| 5755 * buffers as they come in. */ | |
| 5756 if(buffers_processed > 0) | |
| 5757 { | |
|
2
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1
diff
changeset
|
5758 ALint temp_count; |
| 0 | 5759 /* Do as a for-loop because I don't want |
| 5760 * to have to create an array for the | |
| 5761 * unqueued_buffer_id's | |
| 5762 */ | |
| 5763 for(temp_count=0; temp_count<buffers_processed; temp_count++) | |
| 5764 { | |
|
6
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3
diff
changeset
|
5765 /* |
| 0 | 5766 fprintf(stderr, "unqueuing remainder, %d\n", temp_count); |
|
6
4b1048af7e55
Disabled some of the debugging printfs
Eric Wing <ewing . public |-at-| gmail . com>
parents:
3
diff
changeset
|
5767 */ |
| 0 | 5768 alSourceUnqueueBuffers( |
| 5769 ALmixer_Channel_List[i].alsource, | |
| 5770 1, &unqueued_buffer_id | |
| 5771 ); | |
| 5772 if((error = alGetError()) != AL_NO_ERROR) | |
| 5773 { | |
| 5774 fprintf(stderr, "59Testing error: %s\n", | |
|
2
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Eric Wing <ewing . public |-at-| gmail . com>
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1
diff
changeset
|
5775 alGetString(error)); |
| 0 | 5776 } |
| 5777 } | |
|
6
4b1048af7e55
Disabled some of the debugging printfs
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parents:
3
diff
changeset
|
5778 /* |
| 0 | 5779 fprintf(stderr, "done unqueuing remainder for this loop, %d\n", temp_count); |
|
6
4b1048af7e55
Disabled some of the debugging printfs
Eric Wing <ewing . public |-at-| gmail . com>
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3
diff
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|
5780 */ |
|
4b1048af7e55
Disabled some of the debugging printfs
Eric Wing <ewing . public |-at-| gmail . com>
parents:
3
diff
changeset
|
5781 |
| 0 | 5782 /* Need to update counts since we removed everything. |
| 5783 * If we don't update the counts here, we end up in the | |
| 5784 * "Shouldn't be here section, but maybe it's okay due to race conditions" | |
| 5785 */ | |
| 5786 alGetSourcei( | |
| 5787 ALmixer_Channel_List[i].alsource, | |
| 5788 AL_BUFFERS_QUEUED, &buffers_still_queued | |
| 5789 ); | |
| 5790 if((error = alGetError()) != AL_NO_ERROR) | |
| 5791 { | |
| 5792 fprintf(stderr, "5100Testing error: %s\n", | |
|
2
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Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
5793 alGetString(error)); |
| 0 | 5794 } |
| 5795 /* Get the number of buffers processed | |
| 5796 * so we know if we need to refill | |
| 5797 */ | |
| 5798 alGetSourcei( | |
| 5799 ALmixer_Channel_List[i].alsource, | |
| 5800 AL_BUFFERS_PROCESSED, &buffers_processed | |
| 5801 ); | |
| 5802 if((error = alGetError()) != AL_NO_ERROR) | |
| 5803 { | |
| 5804 fprintf(stderr, "5200Testing error: %s\n", | |
|
2
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
5805 alGetString(error)); |
| 0 | 5806 } |
| 5807 } | |
| 5808 | |
| 5809 | |
| 5810 /* Else if buffers_processed == 0 | |
| 5811 * and buffers_still_queued == 0. | |
| 5812 * then we check to see if the source | |
| 5813 * is still playing. Quit if stopped | |
| 5814 * We shouldn't need to worry about | |
| 5815 * looping because that should have | |
| 5816 * been handled above. | |
| 5817 */ | |
| 5818 if(0 == buffers_still_queued) | |
| 5819 { | |
| 5820 /* Make sure playback has stopped before | |
| 5821 * we shutdown. | |
| 5822 */ | |
| 5823 alGetSourcei( | |
| 5824 ALmixer_Channel_List[i].alsource, | |
| 5825 AL_SOURCE_STATE, &state | |
| 5826 ); | |
| 5827 if((error = alGetError()) != AL_NO_ERROR) | |
| 5828 { | |
| 5829 fprintf(stderr, "60Testing error: %s\n", | |
|
2
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
5830 alGetString(error)); |
| 0 | 5831 } |
| 5832 if(AL_STOPPED == state) | |
| 5833 { | |
| 5834 ALmixer_Channel_List[i].almixer_data->num_buffers_in_use = 0; | |
| 5835 /* Playback has ended. | |
| 5836 * Loop if necessary, or launch callback | |
| 5837 * and clear channel (or clear channel and | |
| 5838 * then launch callback?) | |
|
10
c808684660a7
Bug fix: Moved Invoke_Callback before CleanChannel because I was trying to get the ALmixer_Data pointer, but it was cleared:
Eric Wing <ewing . public |-at-| gmail . com>
parents:
6
diff
changeset
|
5839 * Update: Need to do callback first because I reference the mixer_data and source |
| 0 | 5840 */ |
|
10
c808684660a7
Bug fix: Moved Invoke_Callback before CleanChannel because I was trying to get the ALmixer_Data pointer, but it was cleared:
Eric Wing <ewing . public |-at-| gmail . com>
parents:
6
diff
changeset
|
5841 |
|
c808684660a7
Bug fix: Moved Invoke_Callback before CleanChannel because I was trying to get the ALmixer_Data pointer, but it was cleared:
Eric Wing <ewing . public |-at-| gmail . com>
parents:
6
diff
changeset
|
5842 /* Launch callback */ |
|
c808684660a7
Bug fix: Moved Invoke_Callback before CleanChannel because I was trying to get the ALmixer_Data pointer, but it was cleared:
Eric Wing <ewing . public |-at-| gmail . com>
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6
diff
changeset
|
5843 Invoke_Channel_Done_Callback(i, AL_TRUE); |
|
c808684660a7
Bug fix: Moved Invoke_Callback before CleanChannel because I was trying to get the ALmixer_Data pointer, but it was cleared:
Eric Wing <ewing . public |-at-| gmail . com>
parents:
6
diff
changeset
|
5844 |
| 0 | 5845 Clean_Channel(i); |
| 5846 /* Subtract counter */ | |
| 5847 Is_Playing_global--; | |
| 5848 | |
| 5849 | |
| 5850 /* We're done for this loop. | |
| 5851 * Go to next channel | |
| 5852 */ | |
| 5853 continue; | |
| 5854 } | |
| 5855 } /* End end-playback */ | |
| 5856 else | |
| 5857 { | |
| 5858 /* Need to run out buffer */ | |
| 5859 #if 1 | |
| 5860 /* Might want to check state */ | |
| 5861 /* In case the playback stopped, | |
| 5862 * we need to resume */ | |
| 5863 alGetSourcei( | |
| 5864 ALmixer_Channel_List[i].alsource, | |
| 5865 AL_SOURCE_STATE, &state | |
| 5866 ); | |
| 5867 if((error = alGetError()) != AL_NO_ERROR) | |
| 5868 { | |
| 5869 fprintf(stderr, "61Testing error: %s\n", | |
|
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diff
changeset
|
5870 alGetString(error)); |
| 0 | 5871 } |
| 5872 if(AL_STOPPED == state) | |
| 5873 { | |
|
6
4b1048af7e55
Disabled some of the debugging printfs
Eric Wing <ewing . public |-at-| gmail . com>
parents:
3
diff
changeset
|
5874 /* |
| 0 | 5875 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); |
|
6
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3
diff
changeset
|
5876 */ |
| 1 | 5877 /* |
| 0 | 5878 PrintQueueStatus(ALmixer_Channel_List[i].alsource); |
| 1 | 5879 */ |
| 0 | 5880 /* Rather than force unqueuing the buffer, let's see if |
| 5881 * setting the buffer to none works (the OpenAL 1.0 | |
| 5882 * Reference Annotation suggests this should work). | |
| 5883 */ | |
| 5884 alSourcei(ALmixer_Channel_List[i].alsource, | |
| 5885 AL_BUFFER, AL_NONE); | |
| 5886 /* | |
| 5887 PrintQueueStatus(ALmixer_Channel_List[i].alsource); | |
| 5888 */ | |
| 5889 /* This doesn't work because in some cases, I think | |
| 5890 * it causes the sound to be replayed | |
| 5891 */ | |
| 5892 /* | |
| 5893 fprintf(stderr, "Resuming in eof (trying to run out buffers\n"); | |
| 5894 alSourcePlay(ALmixer_Channel_List[i].alsource); | |
| 5895 */ | |
| 5896 } | |
| 5897 #endif | |
| 5898 } /* End trap section */ | |
| 5899 } /* End POST-EOF use-up buffer section */ | |
| 5900 } /* END Streamed section */ | |
| 5901 } /* END channel in use */ | |
| 5902 } /* END for-loop for each channel */ | |
| 5903 | |
| 5904 #ifdef ENABLE_ALMIXER_ALC_SYNC | |
| 5905 alcProcessContext(alcGetCurrentContext()); | |
| 5906 if((error = alGetError()) != AL_NO_ERROR) | |
| 5907 { | |
| 5908 fprintf(stderr, "62Testing error: %s\n", | |
|
2
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diff
changeset
|
5909 alGetString(error)); |
|
279d0427ef26
Overhaul prep for first public release.
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1
diff
changeset
|
5910 } |
|
279d0427ef26
Overhaul prep for first public release.
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1
diff
changeset
|
5911 #endif |
|
279d0427ef26
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1
diff
changeset
|
5912 |
|
279d0427ef26
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1
diff
changeset
|
5913 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
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parents:
1
diff
changeset
|
5914 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
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parents:
1
diff
changeset
|
5915 #endif |
| 0 | 5916 /* Return the number of errors */ |
| 5917 if(error_flag < 0) | |
| 5918 { | |
| 5919 return error_flag; | |
| 5920 } | |
| 5921 /* Return the number of buffers that were queued */ | |
| 5922 return retval; | |
| 5923 } | |
| 5924 | |
| 5925 #ifdef ENABLE_PARANOID_SIGNEDNESS_CHECK | |
| 5926 /* This is only here so we can call SDL_OpenAudio() */ | |
|
2
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1
diff
changeset
|
5927 static void my_dummy_audio_callback(void* userdata, ALbyte* stream, int len) |
| 0 | 5928 { |
| 5929 } | |
| 5930 #endif | |
| 5931 | |
| 5932 | |
| 5933 | |
| 5934 | |
| 5935 #ifdef ENABLE_ALMIXER_THREADS | |
| 5936 /* We might need threads. We | |
| 5937 * must constantly poll OpenAL to find out | |
| 5938 * if sound is being streamed, if play has | |
| 5939 * ended, etc. Without threads, this must | |
| 5940 * be explicitly done by the user. | |
| 5941 * We could try to do it for them if we | |
| 5942 * finish the threads. | |
| 5943 */ | |
| 5944 | |
| 5945 static int Stream_Data_Thread_Callback(void* data) | |
| 5946 { | |
|
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5947 ALint retval; |
| 0 | 5948 |
| 5949 while(ALmixer_Initialized) | |
| 5950 { | |
| 5951 retval = Update_ALmixer(data); | |
| 5952 /* 0 means that nothing needed updating and | |
| 5953 * the function returned quickly | |
| 5954 */ | |
| 5955 if(0 == retval) | |
| 5956 { | |
| 5957 /* Let's be nice and make the thread sleep since | |
| 5958 * little work was done in update | |
| 5959 */ | |
| 5960 /* Make sure times are multiples of 10 | |
| 5961 * for optimal performance and accuracy in Linux | |
| 5962 */ | |
|
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5963 ALmixer_Delay(10); |
| 0 | 5964 } |
| 5965 else | |
| 5966 { | |
| 5967 /* should I also be sleeping/yielding here? */ | |
|
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5968 ALmixer_Delay(0); |
| 0 | 5969 } |
| 5970 } | |
|
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5971 /* |
| 0 | 5972 fprintf(stderr, "Thread is closing\n"); |
|
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5973 */ |
| 0 | 5974 return 0; |
| 5975 } | |
| 5976 #endif /* End of ENABLE_ALMIXER_THREADS */ | |
| 5977 | |
| 5978 | |
|
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5979 /* SDL/SDL_mixer returns -1 on error and 0 on success. |
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5980 * I actually prefer false/true conventions (SDL_Sound/OpenAL/GL) |
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5981 * so SDL_mixer porting people beware. |
| 0 | 5982 * Warning: SDL_QuitSubSystem(SDL_INIT_AUDIO) is called which |
| 5983 * means the SDL audio system will be disabled. It will not | |
| 5984 * be restored (in case SDL is not actually being used) so | |
| 5985 * the user will need to restart it if they need it after | |
| 5986 * OpenAL shuts down. | |
| 5987 */ | |
|
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5988 ALboolean ALmixer_Init(ALuint frequency, ALint num_sources, ALuint refresh) |
| 0 | 5989 { |
| 5990 ALCdevice* dev; | |
| 5991 ALCcontext* context; | |
|
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5992 ALint i; |
| 0 | 5993 ALenum error; |
| 5994 ALuint* source; | |
| 5995 | |
| 5996 #ifdef USING_LOKI_AL_DIST | |
| 5997 /* The Loki dist requires that I set both the | |
| 5998 * device and context frequency values separately | |
| 5999 */ | |
| 6000 /* Hope this won't overflow */ | |
| 6001 char device_string[256]; | |
| 6002 #endif | |
| 6003 | |
| 6004 /* (Venting frustration) Damn it! Nobody bothered | |
| 6005 * documenting how you're supposed to use an attribute | |
| 6006 * list. In fact, the not even the Loki test program | |
| 6007 * writers seem to know because they use it inconsistently. | |
| 6008 * For example, how do you terminate that attribute list? | |
| 6009 * The Loki test code does it 3 different ways. They | |
| 6010 * set the last value to 0, or they set it to ALC_INVALID, | |
| 6011 * or they set two final values: ALC_INVALID, 0 | |
| 6012 * In Loki, 0 and ALC_INVALID happen to be the same, | |
| 6013 * but with Creative Labs ALC_INVALID is -1. | |
| 6014 * So something's going to break. Loki's source | |
| 6015 * code says to terminate with ALC_INVALID. But I | |
| 6016 * don't know if that's really true, or it happens | |
| 6017 * to be a coinicidence because it's defined to 0. | |
| 6018 * Creative provides no source code, so I can't look at how | |
| 6019 * they terminate it. | |
| 6020 * So this is really, really ticking me off... | |
| 6021 * For now, I'm going to use ALC_INVALID. | |
| 6022 * (Update...after further review of the API spec, | |
| 6023 * it seems that a NULL terminated string is the correct | |
| 6024 * termination value to use, so 0 it is.) | |
| 6025 */ | |
| 6026 #if 0 | |
| 6027 ALint attrlist[] = { | |
| 6028 ALC_FREQUENCY, ALMIXER_DEFAULT_FREQUENCY, | |
| 6029 /* Don't know anything about these values. | |
| 6030 * Trust defaults? */ | |
| 6031 /* Supposed to be the refresh rate in Hz. | |
| 6032 * I think 15-120 are supposed to be good | |
| 6033 * values. Though I haven't gotten any effect except | |
| 6034 * for one strange instance on a Mac. But it was | |
| 6035 * unrepeatable. | |
| 6036 */ | |
| 6037 #if 0 | |
| 6038 ALC_REFRESH, 15, | |
| 6039 #endif | |
| 6040 /* Sync requires a alcProcessContext() call | |
| 6041 * for every cycle. By default, this is | |
| 6042 * not used and the value is AL_FALSE | |
| 6043 * because it will probably perform | |
| 6044 * pretty badly for me. | |
| 6045 */ | |
| 6046 #ifdef ENABLE_ALMIXER_ALC_SYNC | |
| 6047 ALC_SYNC, AL_TRUE, | |
| 6048 #else | |
| 6049 ALC_SYNC, AL_FALSE, | |
| 6050 #endif | |
| 6051 /* Looking at the API spec, it implies | |
| 6052 * that the list be a NULL terminated string | |
| 6053 * so it's probably not safe to use ALC_INVALID | |
| 6054 */ | |
| 6055 /* | |
| 6056 ALC_INVALID }; | |
| 6057 */ | |
| 6058 '\0'}; | |
| 6059 #endif | |
| 6060 /* Redo: I'm going to allow ALC_REFRESH to be set. | |
| 6061 * However, if no value is specified, I don't | |
| 6062 * want it in the list so I can get the OpenAL defaults | |
| 6063 */ | |
| 6064 ALint attrlist[7]; | |
|
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6065 ALsizei current_attrlist_index = 0; |
| 0 | 6066 |
| 6067 #ifdef ENABLE_PARANOID_SIGNEDNESS_CHECK | |
| 6068 /* More problems: I'm getting bit by endian/signedness issues on | |
| 6069 * different platforms. I can find the endianess easily enough, | |
| 6070 * but I don't know how to determine what the correct signedness | |
| 6071 * is (if such a thing exists). I do know that if I try using | |
| 6072 * unsigned on OSX with an originally signed sample, I get | |
| 6073 * distortion. However, I don't have any native unsigned samples | |
| 6074 * to test. But I'm assuming that the platform must be in the | |
| 6075 * correct signedness no matter what. | |
| 6076 * I can either assume everybody is signed, or I can try to | |
| 6077 * determine the value. If I try to determine the values, | |
| 6078 * I think my only ability to figure it out will be to open | |
| 6079 * SDL_Audio, and read what the obtained settings were. | |
| 6080 * Then shutdown everything. However, I don't even know how | |
| 6081 * reliable this is. | |
| 6082 * Update: I think I resolved the issues...forgot to update | |
| 6083 * these comments when it happened. I should check the revision control | |
| 6084 * log... Anyway, I think the issue was partly related to me not | |
| 6085 * doing something correctly with the AudioInfo or some kind | |
| 6086 * of stupid endian bug in my code, and weirdness ensued. Looking at the | |
| 6087 * revision control, I think I might have assumed that SDL_Sound would | |
| 6088 * do the right thing with a NULL AudioInfo, but I was incorrect, | |
| 6089 * and had to fill one out myself. | |
| 6090 */ | |
| 6091 SDL_AudioSpec desired; | |
| 6092 SDL_AudioSpec obtained; | |
| 6093 #endif | |
| 6094 | |
| 6095 | |
| 6096 /* Make sure ALmixer isn't already initialized */ | |
| 6097 if(ALmixer_Initialized) | |
| 6098 { | |
|
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6099 return AL_FALSE; |
| 0 | 6100 } |
| 6101 | |
|
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6102 #ifdef ALMIXER_COMPILE_WITHOUT_SDL |
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6103 ALmixer_InitTime(); |
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6104 |
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6105 /* Note: The pool may have been created on previous Init's */ |
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6106 /* I leave the pool allocated allocated in case the user wants |
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6107 * to read the pool in case of a failure (such as in this function). |
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6108 * This is not actually a leak. |
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6109 */ |
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6110 if(NULL == s_ALmixerErrorPool) |
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6111 { |
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6112 s_ALmixerErrorPool = TError_CreateErrorPool(); |
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6113 } |
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6114 if(NULL == s_ALmixerErrorPool) |
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6115 { |
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6116 return AL_FALSE; |
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6117 } |
|
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6118 /* |
|
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6119 fprintf(stderr, "tError Test0\n"); |
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6120 ALmixer_SetError("Initing (and testing SetError)"); |
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6121 fprintf(stderr, "tError Test1: %s\n", ALmixer_GetError()); |
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6122 fprintf(stderr, "tError Test2: %s\n", ALmixer_GetError()); |
|
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6123 */ |
|
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6124 #endif |
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6125 |
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6126 |
| 0 | 6127 /* Set the defaults */ |
|
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6128 /* |
| 0 | 6129 attrlist[0] = ALC_FREQUENCY; |
| 6130 attrlist[1] = ALMIXER_DEFAULT_FREQUENCY; | |
| 6131 attrlist[2] = ALC_SYNC; | |
| 6132 #ifdef ENABLE_ALMIXER_ALC_SYNC | |
| 6133 attrlist[3] = ALC_TRUE; | |
| 6134 #else | |
| 6135 attrlist[3] = ALC_FALSE; | |
| 6136 #endif | |
|
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6137 */ |
| 0 | 6138 /* Set frequency value if it is not 0 */ |
| 6139 if(0 != frequency) | |
| 6140 { | |
|
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6141 attrlist[current_attrlist_index] = ALC_FREQUENCY; |
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6142 current_attrlist_index++; |
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6143 attrlist[current_attrlist_index] = (ALint)frequency; |
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6144 current_attrlist_index++; |
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6145 } |
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6146 |
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6147 #ifdef ENABLE_ALMIXER_ALC_SYNC |
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6148 attrlist[current_attrlist_index] = ALC_SYNC; |
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6149 current_attrlist_index++; |
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6150 attrlist[current_attrlist_index] = ALC_TRUE; |
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6151 current_attrlist_index++; |
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6152 #endif |
| 0 | 6153 |
| 6154 /* If the user specifies a refresh value, | |
| 6155 * make room for it | |
| 6156 */ | |
| 6157 if(0 != refresh) | |
| 6158 { | |
|
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6159 attrlist[current_attrlist_index] = (ALint)ALC_REFRESH; |
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6160 current_attrlist_index++; |
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6161 attrlist[current_attrlist_index] = refresh; |
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6162 current_attrlist_index++; |
| 0 | 6163 } |
| 6164 | |
|
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6165 /* End attribute list */ |
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6166 attrlist[current_attrlist_index] = '\0'; |
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6167 |
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6168 |
| 0 | 6169 /* Initialize SDL_Sound */ |
| 6170 if(! Sound_Init() ) | |
| 6171 { | |
| 6172 ALmixer_SetError(Sound_GetError()); | |
|
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6173 return AL_FALSE; |
| 0 | 6174 } |
| 6175 #ifdef ENABLE_PARANOID_SIGNEDNESS_CHECK | |
| 6176 /* Here is the paranoid check that opens | |
| 6177 * SDL audio in an attempt to find the correct | |
| 6178 * system values. | |
| 6179 */ | |
| 6180 /* Doesn't have to be the actual value I think | |
| 6181 * (as long as it doesn't influence format, in | |
| 6182 * which case I'm probably screwed anyway because OpenAL | |
| 6183 * may easily choose to do something else). | |
| 6184 */ | |
| 6185 desired.freq = 44100; | |
| 6186 desired.channels = 2; | |
| 6187 desired.format = AUDIO_S16SYS; | |
| 6188 desired.callback = my_dummy_audio_callback; | |
| 6189 if(SDL_OpenAudio(&desired, &obtained) >= 0) | |
| 6190 { | |
| 6191 SIGN_TYPE_16BIT_FORMAT = obtained.format; | |
| 6192 /* Now to get really paranoid, we should probably | |
| 6193 * also assume that the 8bit format is also the | |
| 6194 * same sign type and set that value | |
| 6195 */ | |
| 6196 if(AUDIO_S16SYS == obtained.format) | |
| 6197 { | |
| 6198 SIGN_TYPE_8BIT_FORMAT = AUDIO_S8; | |
| 6199 } | |
| 6200 /* Should be AUDIO_U16SYS */ | |
| 6201 else | |
| 6202 { | |
| 6203 SIGN_TYPE_8BIT_FORMAT = AUDIO_U8; | |
| 6204 } | |
| 6205 SDL_CloseAudio(); | |
| 6206 } | |
| 6207 else | |
| 6208 { | |
| 6209 /* Well, I guess I'm in trouble. I guess it's my best guess | |
| 6210 */ | |
| 6211 SIGN_TYPE_16_BIT_FORMAT = AUDIO_S16SYS; | |
| 6212 SIGN_TYPE_8_BIT_FORMAT = AUDIO_S8; | |
| 6213 } | |
| 6214 #endif | |
|
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6215 |
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6216 #ifndef ALMIXER_COMPILE_WITHOUT_SDL |
| 0 | 6217 /* Weirdness: It seems that SDL_Init(SDL_INIT_AUDIO) |
| 6218 * causes OpenAL and SMPEG to conflict. For some reason | |
| 6219 * if SDL_Init on audio is active, then all the SMPEG | |
| 6220 * decoded sound comes out silent. Unfortunately, | |
| 6221 * Sound_Init() invokes SDL_Init on audio. I'm | |
| 6222 * not sure why it actually needs it... | |
| 6223 * But we'll attempt to disable it here after the | |
| 6224 * SDL_Sound::Init call and hope it doesn't break SDL_Sound. | |
| 6225 */ | |
| 6226 SDL_QuitSubSystem(SDL_INIT_AUDIO); | |
|
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6227 #endif |
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6228 |
| 0 | 6229 /* I'm told NULL will call the default string |
| 6230 * and hopefully do the right thing for each platform | |
| 6231 */ | |
| 6232 /* | |
| 6233 dev = alcOpenDevice( NULL ); | |
| 6234 */ | |
| 6235 /* Now I'm told I need to set both the device and context | |
| 6236 * to have the same sampling rate, so I must pass a string | |
| 6237 * to OpenDevice(). I don't know how portable these strings are. | |
| 6238 * I don't even know if the format for strings is | |
| 6239 * compatible | |
| 6240 * From the testattrib.c in the Loki test section | |
| 6241 * dev = alcOpenDevice( (const ALubyte *) "'((sampling-rate 22050))" ); | |
| 6242 */ | |
| 6243 | |
| 6244 #ifdef USING_LOKI_AL_DIST | |
| 6245 sprintf(device_string, "'((sampling-rate %d))", attrlist[1]); | |
| 6246 dev = alcOpenDevice( (const ALubyte *) device_string ); | |
| 6247 #else | |
| 6248 dev = alcOpenDevice( NULL ); | |
| 6249 #endif | |
| 6250 if(NULL == dev) | |
| 6251 { | |
| 6252 ALmixer_SetError("Cannot open sound device for OpenAL"); | |
|
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6253 return AL_FALSE; |
|
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|
6254 } |
|
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6255 |
|
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|
6256 #ifdef __APPLE__ |
|
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|
6257 /* The ALC_FREQUENCY attribute is ignored with Apple's implementation. */ |
|
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|
6258 /* This extension must be called before the context is created. */ |
|
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6259 if(0 != frequency) |
|
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6260 { |
|
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|
6261 Internal_alcMacOSXMixerOutputRate((ALdouble)frequency); |
|
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6262 } |
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|
6263 ALmixer_Frequency_global = (ALuint)Internal_alcMacOSXGetMixerOutputRate(); |
|
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6264 /* |
|
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|
6265 fprintf(stderr, "Internal_alcMacOSXMixerOutputRate is: %lf", Internal_alcMacOSXGetMixerOutputRate()); |
|
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6266 */ |
|
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|
6267 #endif |
| 0 | 6268 |
| 6269 context = alcCreateContext(dev, attrlist); | |
| 6270 if(NULL == context) | |
| 6271 { | |
| 6272 ALmixer_SetError("Cannot create a context OpenAL"); | |
| 6273 alcCloseDevice(dev); | |
|
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|
6274 return AL_FALSE; |
| 0 | 6275 } |
| 6276 | |
| 6277 | |
| 6278 /* Hmmm, OSX is returning 1 on alcMakeContextCurrent, | |
| 6279 * but ALC_NO_ERROR is defined to ALC_FALSE. | |
| 6280 * According to Garin Hiebert, this is actually an inconsistency | |
| 6281 * in the Loki version. The function should return a boolean. | |
| 6282 * instead of ALC_NO_ERROR. Garin suggested I check via | |
| 6283 * alcGetError(). | |
| 6284 */ | |
| 6285 /* clear the error */ | |
| 6286 alcGetError(dev); | |
| 6287 alcMakeContextCurrent(context); | |
| 6288 | |
| 6289 error = alcGetError(dev); | |
| 6290 if( (ALC_NO_ERROR != error) ) | |
| 6291 { | |
| 6292 ALmixer_SetError("Could not MakeContextCurrent"); | |
| 6293 alcDestroyContext(context); | |
| 6294 alcCloseDevice(dev); | |
|
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6295 return AL_FALSE; |
|
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6296 } |
|
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6297 |
|
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|
6298 /* It looks like OpenAL won't let us ask it what |
|
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|
6299 * the set frequency is, so we need to save our |
|
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6300 * own copy. Yuck. |
|
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|
6301 * Update: J. Valenzuela just updated the Loki |
|
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|
6302 * dist (2003/01/02) to handle this. |
|
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6303 * The demo is in testattrib.c. |
|
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6304 */ |
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6305 /* |
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|
6306 ALmixer_Frequency_global = frequency; |
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6307 */ |
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6308 #ifndef __APPLE__ |
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6309 alcGetIntegerv(dev, ALC_FREQUENCY, 1, &ALmixer_Frequency_global); |
|
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6310 /* |
|
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6311 fprintf(stderr, "alcGetIntegerv ALC_FREQUENCY is: %d", ALmixer_Frequency_global); |
|
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6312 */ |
|
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6313 #endif |
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6314 |
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6315 |
| 0 | 6316 #if 0 |
| 6317 /* OSX is failing on alcMakeContextCurrent(). Try checking it first? */ | |
| 6318 if(alcGetCurrentContext() != context) | |
| 6319 { | |
| 6320 /* Hmmm, OSX is returning 1 on alcMakeContextCurrent, | |
| 6321 * but ALC_NO_ERROR is defined to ALC_FALSE. | |
| 6322 * I think this is a bug in the OpenAL implementation. | |
| 6323 */ | |
| 6324 fprintf(stderr,"alcMakeContextCurrent returns %d\n", alcMakeContextCurrent(context)); | |
| 6325 | |
| 6326 fprintf(stderr, "Making context current\n"); | |
| 6327 #ifndef __APPLE__ | |
| 6328 if(alcMakeContextCurrent(context) != ALC_NO_ERROR) | |
| 6329 #else | |
| 6330 if(!alcMakeContextCurrent(context)) | |
| 6331 #endif | |
| 6332 { | |
| 6333 ALmixer_SetError("Could not MakeContextCurrent"); | |
| 6334 alcDestroyContext(context); | |
| 6335 alcCloseDevice(dev); | |
|
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6336 return AL_FALSE; |
| 0 | 6337 } |
| 6338 } | |
| 6339 #endif | |
| 6340 | |
| 6341 | |
| 6342 /* #endif */ | |
| 6343 /* Saw this in the README with the OS X OpenAL distribution. | |
| 6344 * It looked interesting and simple, so I thought I might | |
| 6345 * try it out. | |
| 6346 * ***** ALC_CONVERT_DATA_UPON_LOADING | |
| 6347 * This extension allows the caller to tell OpenAL to preconvert to the native Core | |
| 6348 * Audio format, the audio data passed to the | |
| 6349 * library with the alBufferData() call. Preconverting the audio data, reduces CPU | |
| 6350 * usage by removing an audio data conversion | |
| 6351 * (per source) at render timem at the expense of a larger memory footprint. | |
| 6352 * | |
| 6353 * This feature is toggled on/off by using the alDisable() & alEnable() APIs. This | |
| 6354 * setting will be applied to all subsequent | |
| 6355 * calls to alBufferData(). | |
| 6356 */ | |
| 6357 #ifdef __APPLE__ | |
|
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6358 /* |
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|
6359 #if (TARGET_OS_IPHONE == 1) || (TARGET_IPHONE_SIMULATOR == 1) |
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6360 |
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6361 #else |
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6362 #endif |
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|
6363 */ |
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|
6364 ALenum convert_data_enum = alcGetEnumValue(dev, "ALC_MAC_OSX_CONVERT_DATA_UPON_LOADING"); |
|
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|
6365 /* |
|
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|
6366 fprintf(stderr, "ALC_MAC_OSX_CONVERT_DATA_UPON_LOADING=0x%x", convert_data_enum); |
|
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|
6367 */ |
|
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|
6368 if(0 != convert_data_enum) |
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6369 { |
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|
6370 alEnable(convert_data_enum); |
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6371 } |
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|
6372 if( (AL_NO_ERROR != alGetError()) ) |
|
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6373 { |
|
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|
6374 fprintf(stderr, "ALC_MAC_OSX_CONVERT_DATA_UPON_LOADING attempted but failed"); |
|
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|
6375 ALmixer_SetError("ALC_MAC_OSX_CONVERT_DATA_UPON_LOADING attempted but failed"); |
|
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6376 } |
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|
6377 |
| 0 | 6378 #endif |
| 6379 | |
| 6380 | |
| 6381 | |
| 6382 | |
| 6383 ALmixer_Initialized = 1; | |
| 6384 | |
| 6385 if(num_sources <= 0) | |
| 6386 { | |
| 6387 Number_of_Channels_global = ALMIXER_DEFAULT_NUM_CHANNELS; | |
| 6388 } | |
| 6389 else | |
| 6390 { | |
| 6391 Number_of_Channels_global = num_sources; | |
| 6392 } | |
| 6393 Number_of_Reserve_Channels_global = 0; | |
| 6394 Is_Playing_global = 0; | |
| 6395 /* Set to Null in case system quit and was reinitialized */ | |
| 6396 Channel_Done_Callback = NULL; | |
| 6397 Channel_Done_Callback_Userdata = NULL; | |
| 6398 Channel_Data_Callback = NULL; | |
| 1 | 6399 Channel_Data_Callback_Userdata = NULL; |
| 0 | 6400 |
| 6401 /* Allocate memory for the list of channels */ | |
| 6402 ALmixer_Channel_List = (struct ALmixer_Channel*) malloc(Number_of_Channels_global * sizeof(struct ALmixer_Channel)); | |
| 6403 if(NULL == ALmixer_Channel_List) | |
| 6404 { | |
| 6405 ALmixer_SetError("Out of Memory for Channel List"); | |
| 6406 alcDestroyContext(context); | |
| 6407 alcCloseDevice(dev); | |
| 6408 ALmixer_Initialized = 0; | |
| 6409 Number_of_Channels_global = 0; | |
|
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|
6410 return AL_FALSE; |
| 0 | 6411 } |
| 6412 | |
| 6413 /* Allocate memory for the list of sources that map to the channels */ | |
| 6414 Source_Map_List = (Source_Map*) malloc(Number_of_Channels_global * sizeof(Source_Map)); | |
| 6415 if(NULL == Source_Map_List) | |
| 6416 { | |
| 6417 ALmixer_SetError("Out of Memory for Source Map List"); | |
| 6418 free(ALmixer_Channel_List); | |
| 6419 alcDestroyContext(context); | |
| 6420 alcCloseDevice(dev); | |
| 6421 ALmixer_Initialized = 0; | |
| 6422 Number_of_Channels_global = 0; | |
|
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6423 return AL_FALSE; |
| 0 | 6424 } |
| 6425 | |
| 6426 /* Create array that will hold the sources */ | |
| 6427 source = (ALuint*)malloc(Number_of_Channels_global * sizeof(ALuint)); | |
| 6428 if(NULL == source) | |
| 6429 { | |
| 6430 ALmixer_SetError("Out of Memory for sources"); | |
| 6431 free(Source_Map_List); | |
| 6432 free(ALmixer_Channel_List); | |
| 6433 alcDestroyContext(context); | |
| 6434 alcCloseDevice(dev); | |
| 6435 ALmixer_Initialized = 0; | |
| 6436 Number_of_Channels_global = 0; | |
|
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6437 return AL_FALSE; |
| 0 | 6438 } |
| 6439 | |
| 6440 /* Clear the error state */ | |
| 6441 alGetError(); | |
| 6442 /* Generate the OpenAL sources */ | |
| 6443 alGenSources(Number_of_Channels_global, source); | |
| 6444 if( (error=alGetError()) != AL_NO_ERROR) | |
| 6445 { | |
|
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|
6446 ALmixer_SetError("Couldn't generate sources: %s\n", alGetString(error)); |
| 0 | 6447 free(ALmixer_Channel_List); |
| 6448 free(Source_Map_List); | |
| 6449 alcDestroyContext(context); | |
| 6450 alcCloseDevice(dev); | |
| 6451 ALmixer_Initialized = 0; | |
| 6452 Number_of_Channels_global = 0; | |
|
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6453 return AL_FALSE; |
| 0 | 6454 } |
| 6455 | |
| 6456 /* Initialize each channel and associate one source to one channel */ | |
| 6457 for(i=0; i<Number_of_Channels_global; i++) | |
| 6458 { | |
| 6459 if(0 == source[i]) | |
| 6460 { | |
| 6461 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"); | |
| 6462 } | |
| 6463 | |
| 6464 Init_Channel(i); | |
| 6465 /* Keeping the source allocation out of the Init function | |
| 6466 * in case I want to reuse the Init | |
| 6467 * function for resetting data | |
| 6468 */ | |
| 6469 ALmixer_Channel_List[i].alsource = source[i]; | |
| 6470 /* Now also keep a copy of the source to channel mapping | |
| 6471 * in case we need to look up a channel from the source | |
| 6472 * instead of a source from a channel | |
| 6473 */ | |
| 6474 Source_Map_List[i].source = source[i]; | |
| 6475 Source_Map_List[i].channel = i; | |
| 6476 /* Clean the channel because there are some things that need to | |
| 6477 * be done that can't happen until the source is set | |
| 6478 */ | |
| 6479 Clean_Channel(i); | |
| 6480 } | |
| 6481 | |
| 6482 /* The Source_Map_List must be sorted by source for binary searches | |
| 6483 */ | |
| 6484 qsort(Source_Map_List, Number_of_Channels_global, sizeof(Source_Map), Compare_Source_Map); | |
| 6485 | |
|
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6486 /* |
|
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6487 ALmixer_OutputDecoders(); |
|
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6488 */ |
|
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6489 #ifdef ENABLE_ALMIXER_THREADS |
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6490 s_simpleLock = SDL_CreateMutex(); |
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|
6491 if(NULL == s_simpleLock) |
| 0 | 6492 { |
| 6493 /* SDL sets the error message already? */ | |
| 6494 free(source); | |
| 6495 free(ALmixer_Channel_List); | |
| 6496 free(Source_Map_List); | |
| 6497 alcDestroyContext(context); | |
| 6498 alcCloseDevice(dev); | |
| 6499 ALmixer_Initialized = 0; | |
| 6500 Number_of_Channels_global = 0; | |
|
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6501 return AL_FALSE; |
| 0 | 6502 } |
| 6503 | |
| 6504 | |
| 6505 Stream_Thread_global = SDL_CreateThread(Stream_Data_Thread_Callback, NULL); | |
| 6506 if(NULL == Stream_Thread_global) | |
| 6507 { | |
| 6508 /* SDL sets the error message already? */ | |
|
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|
6509 SDL_DestroyMutex(s_simpleLock); |
| 0 | 6510 free(source); |
| 6511 free(ALmixer_Channel_List); | |
| 6512 free(Source_Map_List); | |
| 6513 alcDestroyContext(context); | |
| 6514 alcCloseDevice(dev); | |
| 6515 ALmixer_Initialized = 0; | |
| 6516 Number_of_Channels_global = 0; | |
|
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6517 return AL_FALSE; |
| 0 | 6518 } |
| 6519 | |
|
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|
6520 /* |
| 0 | 6521 fprintf(stderr, "Using threads\n"); |
|
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|
6522 */ |
| 0 | 6523 #endif /* End of ENABLE_ALMIXER_THREADS */ |
| 6524 | |
| 6525 /* We don't need this array any more because all the sources | |
| 6526 * are connected to channels | |
| 6527 */ | |
| 6528 free(source); | |
|
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|
6529 return AL_TRUE; |
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|
6530 } |
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|
6531 |
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6532 |
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6533 ALboolean ALmixer_InitContext(ALuint frequency, ALuint refresh) |
| 0 | 6534 { |
| 6535 ALCdevice* dev; | |
| 6536 ALCcontext* context; | |
| 6537 ALCenum error; | |
| 6538 | |
| 6539 #ifdef USING_LOKI_AL_DIST | |
| 6540 /* The Loki dist requires that I set both the | |
| 6541 * device and context frequency values separately | |
| 6542 */ | |
| 6543 /* Hope this won't overflow */ | |
| 6544 char device_string[256]; | |
| 6545 #endif | |
| 6546 | |
| 6547 /* (Venting frustration) Damn it! Nobody bothered | |
| 6548 * documenting how you're supposed to use an attribute | |
| 6549 * list. In fact, the not even the Loki test program | |
| 6550 * writers seem to know because they use it inconsistently. | |
| 6551 * For example, how do you terminate that attribute list? | |
| 6552 * The Loki test code does it 3 different ways. They | |
| 6553 * set the last value to 0, or they set it to ALC_INVALID, | |
| 6554 * or they set two final values: ALC_INVALID, 0 | |
| 6555 * In Loki, 0 and ALC_INVALID happen to be the same, | |
| 6556 * but with Creative Labs ALC_INVALID is -1. | |
| 6557 * So something's going to break. Loki's source | |
| 6558 * code says to terminate with ALC_INVALID. But I | |
| 6559 * don't know if that's really true, or it happens | |
| 6560 * to be a coinicidence because it's defined to 0. | |
| 6561 * Creative provides no source code, so I can't look at how | |
| 6562 * they terminate it. | |
| 6563 * So this is really, really ticking me off... | |
| 6564 * For now, I'm going to use ALC_INVALID. | |
| 6565 * (Update...after further review of the API spec, | |
| 6566 * it seems that a NULL terminated string is the correct | |
| 6567 * termination value to use, so 0 it is.) | |
| 6568 */ | |
| 6569 #if 0 | |
| 6570 ALint attrlist[] = { | |
| 6571 ALC_FREQUENCY, ALMIXER_DEFAULT_FREQUENCY, | |
| 6572 /* Don't know anything about these values. | |
| 6573 * Trust defaults? */ | |
| 6574 /* Supposed to be the refresh rate in Hz. | |
| 6575 * I think 15-120 are supposed to be good | |
| 6576 * values. Though I haven't gotten any effect except | |
| 6577 * for one strange instance on a Mac. But it was | |
| 6578 * unrepeatable. | |
| 6579 */ | |
| 6580 #if 0 | |
| 6581 ALC_REFRESH, 15, | |
| 6582 #endif | |
| 6583 /* Sync requires a alcProcessContext() call | |
| 6584 * for every cycle. By default, this is | |
| 6585 * not used and the value is AL_FALSE | |
| 6586 * because it will probably perform | |
| 6587 * pretty badly for me. | |
| 6588 */ | |
| 6589 #ifdef ENABLE_ALMIXER_ALC_SYNC | |
| 6590 ALC_SYNC, AL_TRUE, | |
| 6591 #else | |
| 6592 ALC_SYNC, AL_FALSE, | |
| 6593 #endif | |
| 6594 /* Looking at the API spec, it implies | |
| 6595 * that the list be a NULL terminated string | |
| 6596 * so it's probably not safe to use ALC_INVALID | |
| 6597 */ | |
| 6598 /* | |
| 6599 ALC_INVALID }; | |
| 6600 */ | |
| 6601 '\0'}; | |
| 6602 #endif | |
| 6603 /* Redo: I'm going to allow ALC_REFRESH to be set. | |
| 6604 * However, if no value is specified, I don't | |
| 6605 * want it in the list so I can get the OpenAL defaults | |
| 6606 */ | |
| 6607 ALint attrlist[7]; | |
|
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6608 ALsizei current_attrlist_index = 0; |
| 0 | 6609 |
| 6610 #ifdef ENABLE_PARANOID_SIGNEDNESS_CHECK | |
| 6611 /* More problems: I'm getting bit by endian/signedness issues on | |
| 6612 * different platforms. I can find the endianess easily enough, | |
| 6613 * but I don't know how to determine what the correct signedness | |
| 6614 * is (if such a thing exists). I do know that if I try using | |
| 6615 * unsigned on OSX with an originally signed sample, I get | |
| 6616 * distortion. However, I don't have any native unsigned samples | |
| 6617 * to test. But I'm assuming that the platform must be in the | |
| 6618 * correct signedness no matter what. | |
| 6619 * I can either assume everybody is signed, or I can try to | |
| 6620 * determine the value. If I try to determine the values, | |
| 6621 * I think my only ability to figure it out will be to open | |
| 6622 * SDL_Audio, and read what the obtained settings were. | |
| 6623 * Then shutdown everything. However, I don't even know how | |
| 6624 * reliable this is. | |
| 6625 * Update: I think I resolved the issues...forgot to update | |
| 6626 * these comments when it happened. I should check the revision control | |
| 6627 * log... Anyway, I think the issue was partly related to me not | |
| 6628 * doing something correctly with the AudioInfo or some kind | |
| 6629 * of stupid endian bug in my code, and weirdness ensued. Looking at the | |
| 6630 * revision control, I think I might have assumed that SDL_Sound would | |
| 6631 * do the right thing with a NULL AudioInfo, but I was incorrect, | |
| 6632 * and had to fill one out myself. | |
| 6633 */ | |
| 6634 SDL_AudioSpec desired; | |
| 6635 SDL_AudioSpec obtained; | |
| 6636 #endif | |
| 6637 | |
| 6638 | |
| 6639 | |
| 6640 | |
| 6641 /* Make sure ALmixer isn't already initialized */ | |
| 6642 if(ALmixer_Initialized) | |
| 6643 { | |
|
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6644 return AL_FALSE; |
| 0 | 6645 } |
| 6646 | |
| 6647 /* Set the defaults */ | |
| 6648 attrlist[0] = ALC_FREQUENCY; | |
| 6649 attrlist[1] = ALMIXER_DEFAULT_FREQUENCY; | |
| 6650 attrlist[2] = ALC_SYNC; | |
| 6651 #ifdef ENABLE_ALMIXER_ALC_SYNC | |
| 6652 attrlist[3] = ALC_TRUE; | |
| 6653 #else | |
| 6654 attrlist[3] = ALC_FALSE; | |
| 6655 #endif | |
| 6656 /* Set frequency value if it is not 0 */ | |
| 6657 if(0 != frequency) | |
| 6658 { | |
|
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|
6659 attrlist[current_attrlist_index] = ALC_FREQUENCY; |
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|
6660 current_attrlist_index++; |
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|
6661 attrlist[current_attrlist_index] = (ALint)frequency; |
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6662 current_attrlist_index++; |
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6663 } |
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6664 |
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6665 #ifdef ENABLE_ALMIXER_ALC_SYNC |
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6666 attrlist[current_attrlist_index] = ALC_SYNC; |
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6667 current_attrlist_index++; |
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6668 attrlist[current_attrlist_index] = ALC_TRUE; |
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6669 current_attrlist_index++; |
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|
6670 #endif |
| 0 | 6671 |
| 6672 /* If the user specifies a refresh value, | |
| 6673 * make room for it | |
| 6674 */ | |
| 6675 if(0 != refresh) | |
| 6676 { | |
|
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6677 attrlist[current_attrlist_index] = (ALint)ALC_REFRESH; |
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6678 current_attrlist_index++; |
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6679 attrlist[current_attrlist_index] = refresh; |
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6680 current_attrlist_index++; |
| 0 | 6681 } |
| 6682 | |
|
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6683 /* End attribute list */ |
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6684 attrlist[current_attrlist_index] = '\0'; |
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6685 |
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6686 |
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6687 |
| 0 | 6688 /* Initialize SDL_Sound */ |
| 6689 if(! Sound_Init() ) | |
| 6690 { | |
| 6691 ALmixer_SetError(Sound_GetError()); | |
|
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6692 return AL_FALSE; |
| 0 | 6693 } |
| 6694 #ifdef ENABLE_PARANOID_SIGNEDNESS_CHECK | |
| 6695 /* Here is the paranoid check that opens | |
| 6696 * SDL audio in an attempt to find the correct | |
| 6697 * system values. | |
| 6698 */ | |
| 6699 /* Doesn't have to be the actual value I think | |
| 6700 * (as long as it doesn't influence format, in | |
| 6701 * which case I'm probably screwed anyway because OpenAL | |
| 6702 * may easily choose to do something else). | |
| 6703 */ | |
| 6704 desired.freq = 44100; | |
| 6705 desired.channels = 2; | |
| 6706 desired.format = AUDIO_S16SYS; | |
| 6707 desired.callback = my_dummy_audio_callback; | |
| 6708 if(SDL_OpenAudio(&desired, &obtained) >= 0) | |
| 6709 { | |
| 6710 SIGN_TYPE_16BIT_FORMAT = obtained.format; | |
| 6711 /* Now to get really paranoid, we should probably | |
| 6712 * also assume that the 8bit format is also the | |
| 6713 * same sign type and set that value | |
| 6714 */ | |
| 6715 if(AUDIO_S16SYS == obtained.format) | |
| 6716 { | |
| 6717 SIGN_TYPE_8BIT_FORMAT = AUDIO_S8; | |
| 6718 } | |
| 6719 /* Should be AUDIO_U16SYS */ | |
| 6720 else | |
| 6721 { | |
| 6722 SIGN_TYPE_8BIT_FORMAT = AUDIO_U8; | |
| 6723 } | |
| 6724 SDL_CloseAudio(); | |
| 6725 } | |
| 6726 else | |
| 6727 { | |
| 6728 /* Well, I guess I'm in trouble. I guess it's my best guess | |
| 6729 */ | |
| 6730 SIGN_TYPE_16_BIT_FORMAT = AUDIO_S16SYS; | |
| 6731 SIGN_TYPE_8_BIT_FORMAT = AUDIO_S8; | |
| 6732 } | |
| 6733 #endif | |
|
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|
6734 |
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|
6735 #ifndef ALMIXER_COMPILE_WITHOUT_SDL |
| 0 | 6736 /* Weirdness: It seems that SDL_Init(SDL_INIT_AUDIO) |
| 6737 * causes OpenAL and SMPEG to conflict. For some reason | |
| 6738 * if SDL_Init on audio is active, then all the SMPEG | |
| 6739 * decoded sound comes out silent. Unfortunately, | |
| 6740 * Sound_Init() invokes SDL_Init on audio. I'm | |
| 6741 * not sure why it actually needs it... | |
| 6742 * But we'll attempt to disable it here after the | |
| 6743 * SDL_Sound::Init call and hope it doesn't break SDL_Sound. | |
| 6744 */ | |
| 6745 SDL_QuitSubSystem(SDL_INIT_AUDIO); | |
|
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|
6746 #endif |
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|
6747 |
| 0 | 6748 /* I'm told NULL will call the default string |
| 6749 * and hopefully do the right thing for each platform | |
| 6750 */ | |
| 6751 /* | |
| 6752 dev = alcOpenDevice( NULL ); | |
| 6753 */ | |
| 6754 /* Now I'm told I need to set both the device and context | |
| 6755 * to have the same sampling rate, so I must pass a string | |
| 6756 * to OpenDevice(). I don't know how portable these strings are. | |
| 6757 * I don't even know if the format for strings is | |
| 6758 * compatible | |
| 6759 * From the testattrib.c in the Loki test section | |
| 6760 * dev = alcOpenDevice( (const ALubyte *) "'((sampling-rate 22050))" ); | |
| 6761 */ | |
| 6762 | |
| 6763 #ifdef USING_LOKI_AL_DIST | |
| 6764 sprintf(device_string, "'((sampling-rate %d))", attrlist[1]); | |
| 6765 dev = alcOpenDevice( (const ALubyte *) device_string ); | |
| 6766 #else | |
| 6767 dev = alcOpenDevice( NULL ); | |
| 6768 #endif | |
| 6769 if(NULL == dev) | |
| 6770 { | |
| 6771 ALmixer_SetError("Cannot open sound device for OpenAL"); | |
|
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6772 return AL_FALSE; |
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|
6773 } |
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6774 |
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|
6775 #ifdef __APPLE__ |
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6776 /* The ALC_FREQUENCY attribute is ignored with Apple's implementation. */ |
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|
6777 /* This extension must be called before the context is created. */ |
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|
6778 if(0 != frequency) |
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|
6779 { |
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|
6780 Internal_alcMacOSXMixerOutputRate((ALdouble)frequency); |
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6781 } |
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|
6782 ALmixer_Frequency_global = (ALuint)Internal_alcMacOSXGetMixerOutputRate(); |
|
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6783 /* |
|
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|
6784 fprintf(stderr, "Internal_alcMacOSXMixerOutputRate is: %lf", Internal_alcMacOSXGetMixerOutputRate()); |
|
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6785 */ |
|
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|
6786 #endif |
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|
6787 |
| 0 | 6788 |
| 6789 context = alcCreateContext(dev, attrlist); | |
| 6790 if(NULL == context) | |
| 6791 { | |
| 6792 ALmixer_SetError("Cannot create a context OpenAL"); | |
| 6793 alcCloseDevice(dev); | |
|
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6794 return AL_FALSE; |
| 0 | 6795 } |
| 6796 | |
| 6797 | |
| 6798 /* Hmmm, OSX is returning 1 on alcMakeContextCurrent, | |
| 6799 * but ALC_NO_ERROR is defined to ALC_FALSE. | |
| 6800 * According to Garin Hiebert, this is actually an inconsistency | |
| 6801 * in the Loki version. The function should return a boolean. | |
| 6802 * instead of ALC_NO_ERROR. Garin suggested I check via | |
| 6803 * alcGetError(). | |
| 6804 */ | |
| 6805 /* clear the error */ | |
| 6806 alcGetError(dev); | |
| 6807 alcMakeContextCurrent(context); | |
| 6808 | |
| 6809 error = alcGetError(dev); | |
| 6810 if( (ALC_NO_ERROR != error) ) | |
| 6811 { | |
| 6812 ALmixer_SetError("Could not MakeContextCurrent"); | |
| 6813 alcDestroyContext(context); | |
| 6814 alcCloseDevice(dev); | |
|
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6815 return AL_FALSE; |
| 0 | 6816 } |
| 6817 | |
| 6818 | |
| 6819 #if 0 | |
| 6820 /* OSX is failing on alcMakeContextCurrent(). Try checking it first? */ | |
| 6821 if(alcGetCurrentContext() != context) | |
| 6822 { | |
| 6823 /* Hmmm, OSX is returning 1 on alcMakeContextCurrent, | |
| 6824 * but ALC_NO_ERROR is defined to ALC_FALSE. | |
| 6825 * I think this is a bug in the OpenAL implementation. | |
| 6826 */ | |
| 6827 fprintf(stderr,"alcMakeContextCurrent returns %d\n", alcMakeContextCurrent(context)); | |
| 6828 | |
| 6829 fprintf(stderr, "Making context current\n"); | |
| 6830 #ifndef __APPLE__ | |
| 6831 if(alcMakeContextCurrent(context) != ALC_NO_ERROR) | |
| 6832 #else | |
| 6833 if(!alcMakeContextCurrent(context)) | |
| 6834 #endif | |
| 6835 { | |
| 6836 ALmixer_SetError("Could not MakeContextCurrent"); | |
| 6837 alcDestroyContext(context); | |
| 6838 alcCloseDevice(dev); | |
|
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6839 return AL_FALSE; |
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|
6840 } |
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6841 |
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6842 } |
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6843 #endif |
|
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6844 |
|
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6845 /* It looks like OpenAL won't let us ask it what |
|
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6846 * the set frequency is, so we need to save our |
|
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6847 * own copy. Yuck. |
|
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|
6848 * Update: J. Valenzuela just updated the Loki |
|
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|
6849 * dist (2003/01/02) to handle this. |
|
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6850 * The demo is in testattrib.c. |
|
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6851 */ |
|
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6852 #ifndef __APPLE__ |
|
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6853 alcGetIntegerv(dev, ALC_FREQUENCY, 1, &ALmixer_Frequency_global); |
|
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6854 /* |
|
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6855 fprintf(stderr, "alcGetIntegerv ALC_FREQUENCY is: %d", ALmixer_Frequency_global); |
|
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6856 */ |
|
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|
6857 #endif |
|
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6858 |
|
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6859 |
| 0 | 6860 |
| 6861 /* Saw this in the README with the OS X OpenAL distribution. | |
| 6862 * It looked interesting and simple, so I thought I might | |
| 6863 * try it out. | |
| 6864 * ***** ALC_CONVERT_DATA_UPON_LOADING | |
| 6865 * This extension allows the caller to tell OpenAL to preconvert to the native Core | |
| 6866 * Audio format, the audio data passed to the | |
| 6867 * library with the alBufferData() call. Preconverting the audio data, reduces CPU | |
| 6868 * usage by removing an audio data conversion | |
| 6869 * (per source) at render timem at the expense of a larger memory footprint. | |
| 6870 * | |
| 6871 * This feature is toggled on/off by using the alDisable() & alEnable() APIs. This | |
| 6872 * setting will be applied to all subsequent | |
| 6873 * calls to alBufferData(). | |
| 6874 */ | |
| 6875 #ifdef __APPLE__ | |
|
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6876 /* |
|
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6877 #if (TARGET_OS_IPHONE == 1) || (TARGET_IPHONE_SIMULATOR == 1) |
|
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6878 |
|
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6879 #else |
|
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6880 #endif |
|
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6881 */ |
|
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6882 ALenum convert_data_enum = alcGetEnumValue(dev, "ALC_MAC_OSX_CONVERT_DATA_UPON_LOADING"); |
|
6
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6883 /* |
|
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|
6884 fprintf(stderr, "ALC_MAC_OSX_CONVERT_DATA_UPON_LOADING=0x%x", convert_data_enum); |
|
6
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6885 */ |
|
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6886 if(0 != convert_data_enum) |
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|
6887 { |
|
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|
6888 alEnable(convert_data_enum); |
|
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6889 } |
|
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|
6890 if( (AL_NO_ERROR != alGetError()) ) |
|
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6891 { |
|
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6892 fprintf(stderr, "ALC_MAC_OSX_CONVERT_DATA_UPON_LOADING attempted but failed"); |
|
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6893 ALmixer_SetError("ALC_MAC_OSX_CONVERT_DATA_UPON_LOADING attempted but failed"); |
|
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6894 } |
|
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6895 #endif |
|
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6896 |
|
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6897 return AL_TRUE; |
|
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6898 } |
|
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6899 |
|
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6900 |
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6901 ALboolean ALmixer_InitMixer(ALint num_sources) |
| 0 | 6902 { |
|
2
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|
6903 ALint i; |
| 0 | 6904 ALenum error; |
| 6905 ALuint* source; | |
| 6906 | |
| 6907 | |
| 6908 ALmixer_Initialized = 1; | |
| 6909 | |
|
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6910 |
|
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6911 #ifdef ALMIXER_COMPILE_WITHOUT_SDL |
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6912 ALmixer_InitTime(); |
|
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6913 |
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|
6914 /* Note: The pool may have been created on previous Init's */ |
|
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6915 /* I leave the pool allocated allocated in case the user wants |
|
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6916 * to read the pool in case of a failure (such as in this function). |
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6917 * This is not actually a leak. |
|
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6918 */ |
|
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|
6919 if(NULL == s_ALmixerErrorPool) |
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6920 { |
|
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|
6921 s_ALmixerErrorPool = TError_CreateErrorPool(); |
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6922 } |
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6923 if(NULL == s_ALmixerErrorPool) |
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6924 { |
|
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|
6925 return AL_FALSE; |
|
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6926 } |
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|
6927 /* |
|
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|
6928 fprintf(stderr, "tError Test0\n"); |
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6929 ALmixer_SetError("Initing (and testing SetError)"); |
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|
6930 fprintf(stderr, "tError Test1: %s\n", ALmixer_GetError()); |
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6931 fprintf(stderr, "tError Test2: %s\n", ALmixer_GetError()); |
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6932 */ |
|
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|
6933 #endif |
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|
6934 |
| 0 | 6935 if(num_sources <= 0) |
| 6936 { | |
| 6937 Number_of_Channels_global = ALMIXER_DEFAULT_NUM_CHANNELS; | |
| 6938 } | |
| 6939 else | |
| 6940 { | |
| 6941 Number_of_Channels_global = num_sources; | |
| 6942 } | |
| 6943 Number_of_Reserve_Channels_global = 0; | |
| 6944 Is_Playing_global = 0; | |
| 6945 /* Set to Null in case system quit and was reinitialized */ | |
| 6946 Channel_Done_Callback = NULL; | |
| 6947 Channel_Done_Callback_Userdata = NULL; | |
| 6948 Channel_Data_Callback = NULL; | |
| 1 | 6949 Channel_Data_Callback_Userdata = NULL; |
| 0 | 6950 |
| 6951 /* Allocate memory for the list of channels */ | |
| 6952 ALmixer_Channel_List = (struct ALmixer_Channel*) malloc(Number_of_Channels_global * sizeof(struct ALmixer_Channel)); | |
| 6953 if(NULL == ALmixer_Channel_List) | |
| 6954 { | |
| 6955 ALmixer_SetError("Out of Memory for Channel List"); | |
| 6956 ALmixer_Initialized = 0; | |
| 6957 Number_of_Channels_global = 0; | |
|
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6958 return AL_FALSE; |
| 0 | 6959 } |
| 6960 | |
| 6961 /* Allocate memory for the list of sources that map to the channels */ | |
| 6962 Source_Map_List = (Source_Map*) malloc(Number_of_Channels_global * sizeof(Source_Map)); | |
| 6963 if(NULL == Source_Map_List) | |
| 6964 { | |
| 6965 ALmixer_SetError("Out of Memory for Source Map List"); | |
| 6966 free(ALmixer_Channel_List); | |
| 6967 ALmixer_Initialized = 0; | |
| 6968 Number_of_Channels_global = 0; | |
|
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|
6969 return AL_FALSE; |
| 0 | 6970 } |
| 6971 | |
| 6972 /* Create array that will hold the sources */ | |
| 6973 source = (ALuint*)malloc(Number_of_Channels_global * sizeof(ALuint)); | |
| 6974 if(NULL == source) | |
| 6975 { | |
| 6976 ALmixer_SetError("Out of Memory for sources"); | |
| 6977 free(Source_Map_List); | |
| 6978 free(ALmixer_Channel_List); | |
| 6979 ALmixer_Initialized = 0; | |
| 6980 Number_of_Channels_global = 0; | |
|
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|
6981 return AL_FALSE; |
| 0 | 6982 } |
| 6983 | |
| 6984 /* Clear the error state */ | |
| 6985 alGetError(); | |
| 6986 /* Generate the OpenAL sources */ | |
| 6987 alGenSources(Number_of_Channels_global, source); | |
| 6988 if( (error=alGetError()) != AL_NO_ERROR) | |
| 6989 { | |
|
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|
6990 ALmixer_SetError("Couldn't generate sources: %s\n", alGetString(error)); |
| 0 | 6991 free(ALmixer_Channel_List); |
| 6992 free(Source_Map_List); | |
| 6993 ALmixer_Initialized = 0; | |
| 6994 Number_of_Channels_global = 0; | |
|
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6995 return AL_FALSE; |
| 0 | 6996 } |
| 6997 | |
| 6998 /* Initialize each channel and associate one source to one channel */ | |
| 6999 for(i=0; i<Number_of_Channels_global; i++) | |
| 7000 { | |
| 7001 Init_Channel(i); | |
| 7002 /* Keeping the source allocation out of the Init function | |
| 7003 * in case I want to reuse the Init | |
| 7004 * function for resetting data | |
| 7005 */ | |
| 7006 ALmixer_Channel_List[i].alsource = source[i]; | |
| 7007 /* Now also keep a copy of the source to channel mapping | |
| 7008 * in case we need to look up a channel from the source | |
| 7009 * instead of a source from a channel | |
| 7010 */ | |
| 7011 Source_Map_List[i].source = source[i]; | |
| 7012 Source_Map_List[i].channel = i; | |
| 7013 /* Clean the channel because there are some things that need to | |
| 7014 * be done that can't happen until the source is set | |
| 7015 */ | |
| 7016 Clean_Channel(i); | |
| 7017 } | |
| 7018 | |
| 7019 /* The Source_Map_List must be sorted by source for binary searches | |
| 7020 */ | |
| 7021 qsort(Source_Map_List, Number_of_Channels_global, sizeof(Source_Map), Compare_Source_Map); | |
| 7022 | |
|
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7023 |
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7024 #ifdef ENABLE_ALMIXER_THREADS |
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7025 s_simpleLock = SDL_CreateMutex(); |
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|
7026 if(NULL == s_simpleLock) |
| 0 | 7027 { |
| 7028 /* SDL sets the error message already? */ | |
| 7029 free(source); | |
| 7030 free(ALmixer_Channel_List); | |
| 7031 free(Source_Map_List); | |
| 7032 ALmixer_Initialized = 0; | |
| 7033 Number_of_Channels_global = 0; | |
|
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|
7034 return AL_FALSE; |
| 0 | 7035 } |
| 7036 | |
| 7037 | |
| 7038 Stream_Thread_global = SDL_CreateThread(Stream_Data_Thread_Callback, NULL); | |
| 7039 if(NULL == Stream_Thread_global) | |
| 7040 { | |
| 7041 /* SDL sets the error message already? */ | |
|
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|
7042 SDL_DestroyMutex(s_simpleLock); |
| 0 | 7043 free(source); |
| 7044 free(ALmixer_Channel_List); | |
| 7045 free(Source_Map_List); | |
| 7046 ALmixer_Initialized = 0; | |
| 7047 Number_of_Channels_global = 0; | |
|
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7048 return AL_FALSE; |
| 0 | 7049 } |
| 7050 | |
|
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7051 /* |
| 0 | 7052 fprintf(stderr, "Using threads\n"); |
|
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7053 */ |
| 0 | 7054 #endif /* End of ENABLE_ALMIXER_THREADS */ |
| 7055 | |
| 7056 /* We don't need this array any more because all the sources | |
| 7057 * are connected to channels | |
| 7058 */ | |
| 7059 free(source); | |
|
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7060 return AL_TRUE; |
| 0 | 7061 } |
| 7062 | |
| 7063 | |
| 7064 | |
| 7065 /* Keep the return value void to allow easy use with | |
| 7066 * atexit() | |
| 7067 */ | |
| 7068 void ALmixer_Quit() | |
| 7069 { | |
| 7070 ALCcontext* context; | |
| 7071 ALCdevice* dev; | |
|
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7072 ALint i; |
| 0 | 7073 |
| 7074 if( ! ALmixer_Initialized) | |
| 7075 { | |
| 7076 return; | |
| 7077 } | |
|
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7078 #ifdef ENABLE_ALMIXER_THREADS |
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7079 SDL_LockMutex(s_simpleLock); |
|
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7080 #endif |
| 0 | 7081 /* Shutdown everything before closing context */ |
|
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|
7082 Internal_HaltChannel(-1, AL_FALSE); |
| 0 | 7083 |
| 7084 /* This flag will cause the thread to terminate */ | |
| 7085 ALmixer_Initialized = 0; | |
| 7086 #ifdef ENABLE_ALMIXER_THREADS | |
|
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|
7087 SDL_UnlockMutex(s_simpleLock); |
| 0 | 7088 SDL_WaitThread(Stream_Thread_global, NULL); |
| 7089 | |
|
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|
7090 SDL_DestroyMutex(s_simpleLock); |
|
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|
7091 #endif |
| 0 | 7092 |
| 7093 /* Delete all the OpenAL sources */ | |
| 7094 for(i=0; i<Number_of_Channels_global; i++) | |
| 7095 { | |
| 7096 alDeleteSources(1, &ALmixer_Channel_List[i].alsource); | |
| 7097 } | |
| 7098 /* Delete all the channels */ | |
| 7099 free(ALmixer_Channel_List); | |
| 7100 free(Source_Map_List); | |
| 7101 | |
| 7102 /* Reset the Number_of_Channels just in case somebody | |
| 7103 * tries using a ALmixer function. | |
| 7104 * I probably should put "Initialized" checks everywhere, | |
| 7105 * but I'm too lazy at the moment. | |
| 7106 */ | |
| 7107 Number_of_Channels_global = 0; | |
| 7108 | |
| 7109 context = alcGetCurrentContext(); | |
| 7110 if(NULL == context) | |
| 7111 { | |
| 7112 return; | |
| 7113 } | |
| 7114 /* Need to get the device before I close the context */ | |
| 7115 dev = alcGetContextsDevice(context); | |
| 7116 alcDestroyContext(context); | |
| 7117 | |
| 7118 if(NULL == dev) | |
| 7119 { | |
| 7120 return; | |
| 7121 } | |
| 7122 alcCloseDevice(dev); | |
| 7123 | |
| 7124 Sound_Quit(); | |
| 7125 | |
|
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7126 #ifdef ALMIXER_COMPILE_WITHOUT_SDL |
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7127 /* Remember: ALmixer_SetError/GetError calls will not work while this is gone. */ |
|
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|
7128 TError_FreeErrorPool(s_ALmixerErrorPool); |
|
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7129 s_ALmixerErrorPool = NULL; |
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7130 #endif |
| 0 | 7131 return; |
| 7132 } | |
| 7133 | |
|
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|
7134 ALboolean ALmixer_IsInitialized() |
| 0 | 7135 { |
| 7136 return ALmixer_Initialized; | |
| 7137 } | |
| 7138 | |
|
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7139 ALuint ALmixer_GetFrequency() |
| 0 | 7140 { |
| 7141 return ALmixer_Frequency_global; | |
| 7142 } | |
| 7143 | |
|
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|
7144 const ALmixer_version* ALmixer_GetLinkedVersion() |
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7145 { |
|
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|
7146 static ALmixer_version linked_mixver; |
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7147 ALMIXER_GET_COMPILED_VERSION(&linked_mixver); |
| 0 | 7148 return(&linked_mixver); |
| 7149 } | |
| 7150 | |
|
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7151 #ifdef ALMIXER_COMPILE_WITHOUT_SDL |
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7152 |
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7153 const char* ALmixer_GetError() |
|
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7154 { |
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|
7155 const char* error_string = NULL; |
|
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7156 if(NULL == s_ALmixerErrorPool) |
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7157 { |
|
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|
7158 return "Error: You should not call ALmixer_GetError while ALmixer is not initialized"; |
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7159 } |
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|
7160 error_string = TError_GetLastErrorStr(s_ALmixerErrorPool); |
|
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7161 /* SDL returns empty strings instead of NULL */ |
|
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|
7162 if(NULL == error_string) |
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|
7163 { |
|
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|
7164 return ""; |
|
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7165 } |
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7166 else |
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7167 { |
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|
7168 return error_string; |
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7169 } |
|
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7170 } |
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7171 |
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|
7172 void ALmixer_SetError(const char* err_str, ...) |
|
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7173 { |
|
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|
7174 if(NULL == s_ALmixerErrorPool) |
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|
7175 { |
|
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|
7176 fprintf(stderr, "Error: You should not call ALmixer_SetError while ALmixer is not initialized\n"); |
|
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|
7177 return; |
|
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|
7178 } |
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|
7179 va_list argp; |
|
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|
7180 va_start(argp, err_str); |
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|
7181 // SDL_SetError which I'm emulating has no number parameter. |
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|
7182 TError_SetErrorv(s_ALmixerErrorPool, 1, err_str, argp); |
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|
7183 va_end(argp); |
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|
7184 } |
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|
7185 |
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|
7186 #endif |
| 0 | 7187 |
| 7188 | |
| 7189 | |
| 7190 | |
| 7191 #if 0 | |
|
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|
7192 void ALmixer_OutputAttributes() |
| 0 | 7193 { |
| 7194 ALint num_flags = 0; | |
| 7195 ALint* flags = 0; | |
| 7196 int i; | |
| 7197 ALCdevice* dev = alcGetContextsDevice( alcGetCurrentContext() ); | |
| 7198 | |
| 7199 | |
| 7200 printf("custom context\n"); | |
| 7201 | |
| 7202 alcGetIntegerv(dev, ALC_ATTRIBUTES_SIZE, | |
| 7203 sizeof num_flags, &num_flags ); | |
| 7204 | |
| 7205 printf("Number of Flags: %d\n", num_flags); | |
| 7206 | |
| 7207 if(num_flags) | |
| 7208 { | |
| 7209 flags = malloc(sizeof(num_flags) * sizeof(ALint)); | |
| 7210 | |
| 7211 alcGetIntegerv(dev, ALC_ALL_ATTRIBUTES, | |
| 7212 sizeof num_flags * sizeof(ALint), | |
| 7213 flags ); | |
| 7214 } | |
| 7215 for(i = 0; i < num_flags-1; i += 2) | |
| 7216 { | |
| 7217 printf("key 0x%x : value %d\n", | |
| 7218 flags[i], flags[i+1]); | |
| 7219 } | |
| 7220 free(flags); | |
| 7221 } | |
| 7222 #endif | |
| 7223 | |
| 7224 | |
|
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|
7225 void ALmixer_OutputDecoders() |
| 0 | 7226 { |
| 7227 Sound_Version sound_compile_version; | |
| 7228 Sound_Version sound_link_version; | |
| 7229 | |
| 7230 const Sound_DecoderInfo **rc = Sound_AvailableDecoders(); | |
| 7231 const Sound_DecoderInfo **i; | |
| 7232 const char **ext; | |
| 7233 FILE* stream = stdout; | |
| 7234 | |
| 7235 | |
| 7236 fprintf(stream, "SDL_sound Information:\n"); | |
| 7237 | |
| 7238 SOUND_VERSION(&sound_compile_version); | |
| 7239 fprintf(stream, "\tCompiled with SDL_sound version: %d.%d.%d\n", | |
| 7240 sound_compile_version.major, | |
| 7241 sound_compile_version.minor, | |
| 7242 sound_compile_version.patch); | |
| 7243 | |
| 7244 Sound_GetLinkedVersion(&sound_link_version); | |
| 7245 fprintf(stream, "\tRunning (linked) with SDL_sound version: %d.%d.%d\n", | |
| 7246 sound_link_version.major, | |
| 7247 sound_link_version.minor, | |
| 7248 sound_link_version.patch); | |
| 7249 | |
| 7250 fprintf(stream, "Supported sound formats:\n"); | |
| 7251 if (rc == NULL) | |
| 7252 fprintf(stream, " * Apparently, NONE!\n"); | |
| 7253 else | |
| 7254 { | |
| 7255 for (i = rc; *i != NULL; i++) | |
| 7256 { | |
| 7257 fprintf(stream, " * %s\n", (*i)->description); | |
| 7258 | |
| 7259 for (ext = (*i)->extensions; *ext != NULL; ext++) | |
| 7260 fprintf(stream, " File extension \"%s\"\n", *ext); | |
| 7261 | |
| 7262 fprintf(stream, " Written by %s.\n %s\n\n", | |
| 7263 (*i)->author, (*i)->url); | |
| 7264 } /* for */ | |
| 7265 } /* else */ | |
| 7266 | |
| 7267 fprintf(stream, "\n"); | |
| 7268 } | |
| 7269 | |
|
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|
7270 void ALmixer_OutputOpenALInfo() |
|
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|
7271 { |
|
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|
7272 ALmixer_version mixer_compile_version; |
|
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|
7273 const ALmixer_version * mixer_link_version=ALmixer_GetLinkedVersion(); |
| 0 | 7274 FILE* stream = stdout; |
| 7275 | |
| 7276 fprintf(stream, "OpenAL Information:\n"); | |
| 7277 fprintf(stream, "\tAL_VENDOR: %s\n", alGetString( AL_VENDOR ) ); | |
| 7278 fprintf(stream, "\tAL_VERSION: %s\n", alGetString( AL_VERSION ) ); | |
| 7279 fprintf(stream, "\tAL_RENDERER: %s\n", alGetString( AL_RENDERER ) ); | |
| 7280 fprintf(stream, "\tAL_EXTENSIONS: %s\n", alGetString( AL_EXTENSIONS ) ); | |
| 7281 | |
|
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|
7282 ALMIXER_GET_COMPILED_VERSION(&mixer_compile_version); |
| 0 | 7283 fprintf(stream, "\nSDL_ALmixer Information:\n"); |
| 7284 fprintf(stream, "\tCompiled with SDL_ALmixer version: %d.%d.%d\n", | |
| 7285 mixer_compile_version.major, | |
| 7286 mixer_compile_version.minor, | |
| 7287 mixer_compile_version.patch); | |
| 7288 | |
| 7289 fprintf(stream, "\tRunning (linked) with SDL_ALmixer version: %d.%d.%d\n", | |
| 7290 mixer_link_version->major, | |
| 7291 mixer_link_version->minor, | |
| 7292 mixer_link_version->patch); | |
| 7293 | |
| 7294 fprintf(stream, "\tCompile flags: "); | |
| 7295 #ifdef ENABLE_LOKI_QUEUE_FIX_HACK | |
| 7296 fprintf(stream, "ENABLE_LOKI_QUEUE_FIX_HACK "); | |
| 7297 #endif | |
| 7298 #ifdef ENABLE_ALMIXER_THREADS | |
| 7299 fprintf(stream, "ENABLE_ALMIXER_THREADS "); | |
| 7300 #endif | |
| 7301 #ifdef ENABLE_ALC_SYNC | |
| 7302 fprintf(stream, "ENABLE_ALC_SYNC "); | |
| 7303 #endif | |
| 7304 fprintf(stream, "\n"); | |
| 7305 } | |
| 7306 | |
| 7307 | |
|
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7308 ALint ALmixer_AllocateChannels(ALint numchans) |
|
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7309 { |
|
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|
7310 ALint retval; |
|
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7311 #ifdef ENABLE_ALMIXER_THREADS |
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7312 SDL_LockMutex(s_simpleLock); |
|
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7313 #endif |
| 0 | 7314 retval = Internal_AllocateChannels(numchans); |
|
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7315 #ifdef ENABLE_ALMIXER_THREADS |
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7316 SDL_UnlockMutex(s_simpleLock); |
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7317 #endif |
| 0 | 7318 return retval; |
| 7319 } | |
| 7320 | |
| 7321 | |
|
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7322 ALint ALmixer_ReserveChannels(ALint num) |
|
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7323 { |
|
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|
7324 ALint retval; |
|
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7325 #ifdef ENABLE_ALMIXER_THREADS |
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|
7326 SDL_LockMutex(s_simpleLock); |
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|
7327 #endif |
| 0 | 7328 retval = Internal_ReserveChannels(num); |
|
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|
7329 #ifdef ENABLE_ALMIXER_THREADS |
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7330 SDL_UnlockMutex(s_simpleLock); |
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7331 #endif |
| 0 | 7332 return retval; |
| 7333 } | |
| 7334 | |
| 7335 | |
| 7336 | |
| 7337 | |
|
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|
7338 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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7339 { |
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|
7340 ALuint bytes_decoded; |
| 0 | 7341 ALmixer_Data* ret_data; |
| 7342 ALenum error; | |
| 7343 | |
| 7344 /* Allocate memory */ | |
| 7345 ret_data = (ALmixer_Data *)malloc(sizeof(ALmixer_Data)); | |
| 7346 if (NULL == ret_data) | |
| 7347 { | |
| 7348 ALmixer_SetError("Out of memory"); | |
| 7349 return(NULL); | |
| 7350 } | |
| 7351 | |
| 7352 /* Initialize the data fields */ | |
| 7353 | |
| 7354 /* Set the Sound_Sample pointer */ | |
| 7355 ret_data->sample = sample; | |
| 7356 | |
| 7357 /* Flag the data to note that it is not in use */ | |
| 7358 ret_data->in_use = 0; | |
| 7359 | |
| 7360 /* Initialize remaining flags */ | |
| 7361 ret_data->total_time = -1; | |
| 7362 ret_data->eof = 0; | |
| 7363 | |
| 7364 /* Just initialize */ | |
| 7365 ret_data->num_buffers_in_use = 0; | |
| 7366 | |
| 7367 /* Just initialize */ | |
| 7368 ret_data->total_bytes = 0; | |
| 7369 | |
| 7370 /* Just initialize */ | |
| 7371 ret_data->loaded_bytes = 0; | |
| 7372 | |
| 7373 /* Set the max queue buffers (minimum must be 2) */ | |
| 7374 if(max_queue_buffers < 2) | |
| 7375 { | |
| 7376 max_queue_buffers = ALMIXER_DEFAULT_QUEUE_BUFFERS; | |
| 7377 } | |
| 7378 ret_data->max_queue_buffers = max_queue_buffers; | |
| 7379 /* Set up the start up buffers */ | |
| 7380 if(0 == num_startup_buffers) | |
| 7381 { | |
| 7382 num_startup_buffers = ALMIXER_DEFAULT_STARTUP_BUFFERS; | |
| 7383 } | |
| 7384 /* Make sure start up buffers is less or equal to max_queue_buffers */ | |
| 7385 if(num_startup_buffers > max_queue_buffers) | |
| 7386 { | |
| 7387 num_startup_buffers = max_queue_buffers; | |
| 7388 } | |
| 7389 ret_data->num_startup_buffers = num_startup_buffers; | |
| 7390 | |
| 7391 ret_data->buffer_map_list = NULL; | |
| 7392 ret_data->current_buffer = 0; | |
| 7393 | |
| 7394 ret_data->circular_buffer_queue = NULL; | |
| 7395 | |
| 7396 /* Now decode and load the data into a data chunk */ | |
| 7397 /* Different cases for Streamed and Predecoded | |
| 7398 * Streamed might turn into a predecoded if buffersize | |
| 7399 * is large enough */ | |
|
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|
7400 if(AL_FALSE == decode_mode_is_predecoded) |
| 0 | 7401 { |
| 7402 bytes_decoded = Sound_Decode(sample); | |
| 7403 if(sample->flags & SOUND_SAMPLEFLAG_ERROR) | |
| 7404 { | |
| 7405 ALmixer_SetError(Sound_GetError()); | |
| 7406 Sound_FreeSample(sample); | |
| 7407 free(ret_data); | |
| 7408 return NULL; | |
| 7409 } | |
| 7410 | |
| 7411 /* If no data, return an error */ | |
| 7412 if(0 == bytes_decoded) | |
| 7413 { | |
| 7414 ALmixer_SetError("File has no data"); | |
| 7415 Sound_FreeSample(sample); | |
| 7416 free(ret_data); | |
| 7417 return NULL; | |
| 7418 } | |
| 7419 | |
| 7420 /* Note, currently, my Ogg conservative modifications | |
| 7421 * prevent EOF from being detected in the first read | |
| 7422 * because of the weird packet behavior of ov_read(). | |
| 7423 * The EAGAIN will get set, but not the EOF. | |
| 7424 * I don't know the best way to handle this, | |
| 7425 * so for now, Ogg's can only be explicitly | |
| 7426 * predecoded. | |
| 7427 */ | |
| 7428 | |
| 7429 /* Correction: Since we no longer actually keep the | |
| 7430 * streamed data we read here (we rewind and throw | |
| 7431 * it away, and start over on Play), it is | |
| 7432 * safe to read another chunk to see if we've hit EOF | |
| 7433 */ | |
| 7434 if(sample->flags & SOUND_SAMPLEFLAG_EAGAIN) | |
| 7435 { | |
| 7436 bytes_decoded = Sound_Decode(sample); | |
| 7437 if(sample->flags & SOUND_SAMPLEFLAG_ERROR) | |
| 7438 { | |
| 7439 ALmixer_SetError(Sound_GetError()); | |
| 7440 Sound_FreeSample(sample); | |
| 7441 free(ret_data); | |
| 7442 return NULL; | |
| 7443 } | |
| 7444 } | |
| 7445 | |
| 7446 | |
| 7447 /* If we found an EOF, the entire file was | |
| 7448 * decoded, so we can treat it like one. | |
| 7449 */ | |
| 7450 | |
| 7451 if(sample->flags & SOUND_SAMPLEFLAG_EOF) | |
| 7452 { | |
|
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|
7453 /* |
| 0 | 7454 fprintf(stderr, "We got LUCKY! File is predecoded even though STREAM was requested\n"); |
|
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|
7455 */ |
| 0 | 7456 ret_data->decoded_all = 1; |
| 7457 /* Need to keep this information around for | |
| 7458 * seek and rewind abilities. | |
| 7459 */ | |
| 7460 ret_data->total_bytes = bytes_decoded; | |
| 7461 /* For now, the loaded bytes is the same as total bytes, but | |
| 7462 * this could change during a seek operation | |
| 7463 */ | |
| 7464 ret_data->loaded_bytes = bytes_decoded; | |
| 7465 | |
| 7466 /* Let's compute the total playing time | |
| 7467 * SDL_sound does not yet provide this (we're working on | |
| 7468 * that at the moment...) | |
| 7469 */ | |
| 7470 ret_data->total_time = Compute_Total_Time(&sample->desired, bytes_decoded); | |
| 7471 | |
| 7472 /* Create one element in the buffer array for data for OpanAL */ | |
| 7473 ret_data->buffer = (ALuint*)malloc( sizeof(ALuint) ); | |
| 7474 if(NULL == ret_data->buffer) | |
| 7475 { | |
| 7476 ALmixer_SetError("Out of Memory"); | |
| 7477 Sound_FreeSample(sample); | |
| 7478 free(ret_data); | |
| 7479 return NULL; | |
| 7480 } | |
| 7481 /* Clear the error code */ | |
| 7482 alGetError(); | |
| 7483 /* Now generate an OpenAL buffer using that first element */ | |
| 7484 alGenBuffers(1, ret_data->buffer); | |
| 7485 if( (error = alGetError()) != AL_NO_ERROR) | |
| 7486 { | |
|
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7487 ALmixer_SetError("alGenBuffers failed: %s\n", alGetString(error)); |
| 0 | 7488 Sound_FreeSample(sample); |
| 7489 free(ret_data->buffer); | |
| 7490 free(ret_data); | |
| 7491 return NULL; | |
| 7492 } | |
| 7493 | |
| 7494 | |
| 7495 /* Now copy the data to the OpenAL buffer */ | |
| 7496 /* We can't just set a pointer because the API needs | |
| 7497 * its own copy to assist hardware acceleration */ | |
| 7498 alBufferData(ret_data->buffer[0], | |
| 7499 TranslateFormat(&sample->desired), | |
| 7500 sample->buffer, | |
| 7501 bytes_decoded, | |
| 7502 sample->desired.rate | |
| 7503 ); | |
| 7504 if( (error = alGetError()) != AL_NO_ERROR) | |
| 7505 { | |
|
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7506 ALmixer_SetError("alBufferData failed: %s\n", alGetString(error)); |
| 0 | 7507 Sound_FreeSample(sample); |
| 7508 alDeleteBuffers(1, ret_data->buffer); | |
| 7509 free(ret_data->buffer); | |
| 7510 free(ret_data); | |
| 7511 return NULL; | |
| 7512 } | |
| 7513 | |
| 7514 /* We should be done with the sample since it's all | |
| 7515 * predecoded. So we can free the memory */ | |
| 7516 | |
| 7517 /* Additional notes: | |
| 7518 * We need to keep data around in case Seek() is needed | |
| 7519 * or other Sound_AudioInfo is needed. | |
| 7520 * This can either be done by not deleting the sample, | |
| 7521 * or it can be done by dynamically recreating it | |
| 7522 * when we need it. | |
| 7523 */ | |
| 7524 /* Since OpenAL won't let us retrieve it | |
| 7525 * (aka dynamically), we have to keep the Sample | |
| 7526 * around because since the user requested | |
| 7527 * streamed and we offered predecoded, | |
| 7528 * we don't want to mess up the user who | |
| 7529 * was expecting seek support | |
| 7530 * So Don't Do anything | |
| 7531 */ | |
| 7532 /* | |
| 7533 if(0 == access_data) | |
| 7534 { | |
| 7535 Sound_FreeSample(sample); | |
| 7536 ret_data->sample = NULL; | |
| 7537 } | |
| 7538 */ | |
| 7539 /* Else, We keep a copy of the sample around. | |
| 7540 * so don't do anything. | |
| 7541 */ | |
| 7542 | |
| 7543 #if 0 | |
| 7544 #if defined(DISABLE_PREDECODED_SEEK) | |
| 7545 Sound_FreeSample(sample); | |
| 7546 ret_data->sample = NULL; | |
| 7547 #elif !defined(DISABLE_SEEK_MEMORY_OPTIMIZATION) | |
| 7548 Sound_FreeSample(sample); | |
| 7549 ret_data->sample = NULL; | |
| 7550 #else | |
| 7551 /* We keep a copy of the sample around. | |
| 7552 * so don't do anything. | |
| 7553 */ | |
| 7554 #endif | |
| 7555 #endif | |
| 7556 /* okay we're done here */ | |
| 7557 | |
| 7558 } | |
| 7559 /* Else, we need to stream the data, so we'll | |
| 7560 * create multple buffers for queuing */ | |
| 7561 else | |
| 7562 { | |
|
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|
7563 /* |
| 0 | 7564 fprintf(stderr, "Loading streamed data (not lucky)\n"); |
|
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|
7565 */ |
| 0 | 7566 ret_data->decoded_all = 0; |
| 7567 | |
| 7568 /* This information is for predecoded. | |
| 7569 * Set to 0, since we don't know. | |
| 7570 */ | |
| 7571 ret_data->total_bytes = 0; | |
| 7572 | |
| 7573 /* Create buffers for data | |
| 7574 */ | |
| 7575 ret_data->buffer = (ALuint*)malloc( sizeof(ALuint) * max_queue_buffers); | |
| 7576 if(NULL == ret_data->buffer) | |
| 7577 { | |
| 7578 ALmixer_SetError("Out of Memory"); | |
| 7579 Sound_FreeSample(sample); | |
| 7580 free(ret_data); | |
| 7581 return NULL; | |
| 7582 } | |
| 7583 | |
| 7584 /* Clear the error code */ | |
| 7585 alGetError(); | |
| 7586 /* Now generate an OpenAL buffer using that first element */ | |
| 7587 alGenBuffers(max_queue_buffers, ret_data->buffer); | |
| 7588 if( (error = alGetError()) != AL_NO_ERROR) | |
| 7589 { | |
|
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7590 ALmixer_SetError("alGenBuffers failed: %s\n", alGetString(error)); |
| 0 | 7591 Sound_FreeSample(sample); |
| 7592 free(ret_data->buffer); | |
| 7593 free(ret_data); | |
| 7594 return NULL; | |
| 7595 } | |
| 7596 | |
| 7597 /* Redesign: Okay, because of the unqueuing problems and such, | |
| 7598 * I've decided to redesign where and how queuing is handled. | |
| 7599 * Before, everything was queued up here. However, this | |
| 7600 * placed a penalty on load and made performance inconsistent | |
| 7601 * when samples had to be rewound. It did make things easier | |
| 7602 * to queue because I could let OpenAL decide which buffer | |
| 7603 * needed to be queued next. | |
| 7604 * Now, I'm going to push off the queuing to the actual | |
| 7605 * Play() command. I'm going to add some book keeping, | |
| 7606 * and allow for additional buffers to be filled at later | |
| 7607 * times. | |
| 7608 */ | |
| 7609 | |
| 7610 | |
| 7611 /* So first of all, because of I already decoded the sample | |
| 7612 * for testing, I need to decide what to do with it. | |
| 7613 * The best thing would be be to alBufferData() it. | |
| 7614 * The problem is it may conflict with the rest of | |
| 7615 * the system because everything now assumes buffers | |
| 7616 * are entirely stripped (because of the unqueing | |
| 7617 * problem). | |
| 7618 * So it looks like I have to do the crappy thing | |
| 7619 * and throw away the data, and rewind. | |
| 7620 */ | |
| 7621 | |
| 7622 if(0 == Sound_Rewind(ret_data->sample)) | |
| 7623 { | |
| 7624 ALmixer_SetError("Cannot use sample for streamed data because it must be rewindable: %s", Sound_GetError() ); | |
| 7625 Sound_FreeSample(sample); | |
| 7626 free(ret_data->buffer); | |
| 7627 free(ret_data); | |
| 7628 return NULL; | |
| 7629 } | |
| 7630 | |
| 7631 | |
| 7632 /* If the user has selected access_data, we need to | |
| 7633 * keep copies of the queuing buffers around because | |
| 7634 * OpenAL won't let us access the data. | |
| 7635 * Allocate the memory for the buffers here | |
| 7636 * and initialize the albuffer-index map | |
| 7637 */ | |
| 7638 if(access_data) | |
| 7639 { | |
|
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|
7640 ALuint j; |
| 0 | 7641 /* Create buffers for data access |
| 7642 * Should be the same number as the number of queue buffers | |
| 7643 */ | |
| 1 | 7644 ret_data->buffer_map_list = (ALmixer_Buffer_Map*)malloc( sizeof(ALmixer_Buffer_Map) * max_queue_buffers); |
| 0 | 7645 if(NULL == ret_data->buffer_map_list) |
| 7646 { | |
| 7647 ALmixer_SetError("Out of Memory"); | |
| 7648 Sound_FreeSample(sample); | |
| 7649 free(ret_data->buffer); | |
| 7650 free(ret_data); | |
| 7651 return NULL; | |
| 7652 } | |
| 7653 | |
| 7654 ret_data->circular_buffer_queue = CircularQueueUnsignedInt_CreateQueue(max_queue_buffers); | |
| 7655 if(NULL == ret_data->circular_buffer_queue) | |
| 7656 { | |
| 7657 ALmixer_SetError("Out of Memory"); | |
| 7658 free(ret_data->buffer_map_list); | |
| 7659 Sound_FreeSample(sample); | |
| 7660 free(ret_data->buffer); | |
| 7661 free(ret_data); | |
| 7662 return NULL; | |
| 7663 } | |
| 7664 | |
| 7665 | |
| 7666 for(j=0; j<max_queue_buffers; j++) | |
| 7667 { | |
| 7668 ret_data->buffer_map_list[j].albuffer = ret_data->buffer[j]; | |
| 7669 ret_data->buffer_map_list[j].index = j; | |
| 7670 ret_data->buffer_map_list[j].num_bytes = 0; | |
|
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|
7671 ret_data->buffer_map_list[j].data = (ALbyte*)malloc( sizeof(ALbyte) * buffersize); |
| 0 | 7672 if(NULL == ret_data->buffer_map_list[j].data) |
| 7673 { | |
| 7674 ALmixer_SetError("Out of Memory"); | |
| 7675 break; | |
| 7676 } | |
| 7677 } | |
| 7678 /* If an error happened, we have to clean up the memory */ | |
| 7679 if(j < max_queue_buffers) | |
| 7680 { | |
| 7681 fprintf(stderr, "################## Buffer allocation failed\n"); | |
| 7682 for( ; j>=0; j--) | |
| 7683 { | |
| 7684 free(ret_data->buffer_map_list[j].data); | |
| 7685 } | |
| 7686 free(ret_data->buffer_map_list); | |
| 7687 CircularQueueUnsignedInt_FreeQueue(ret_data->circular_buffer_queue); | |
| 7688 Sound_FreeSample(sample); | |
| 7689 free(ret_data->buffer); | |
| 7690 free(ret_data); | |
| 7691 return NULL; | |
| 7692 } | |
| 7693 | |
| 7694 /* The Buffer_Map_List must be sorted by albuffer for binary searches | |
| 7695 */ | |
| 1 | 7696 qsort(ret_data->buffer_map_list, max_queue_buffers, sizeof(ALmixer_Buffer_Map), Compare_Buffer_Map); |
| 0 | 7697 } /* End if access_data==true */ |
| 7698 | |
| 7699 | |
| 7700 } /* End of do stream */ | |
| 7701 } /* end of DECODE_STREAM */ | |
| 7702 /* User requested decode all (easy, nothing to figure out) */ | |
|
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7703 else if(AL_TRUE == decode_mode_is_predecoded) |
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7704 { |
|
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|
7705 #ifdef ALMIXER_DISABLE_PREDECODED_PRECOMPUTE_BUFFER_SIZE_OPTIMIZATION |
|
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7706 /* SDL_sound (behind the scenes) seems to loop on buffer_size chunks |
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7707 * until the buffer is filled. It seems like we can |
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7708 * do much better and precompute the size of the buffer |
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7709 * so looping isn't needed. |
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|
7710 * WARNING: Due to the way SDL_sound is currently implemented, |
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7711 * this may waste a lot of memory up front. |
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7712 * SDL_sound seems to pre-create a buffer of the requested size, |
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7713 * but on DecodeAll, an entirely new buffer is created and |
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7714 * everything is memcpy'd into the new buffer in read chunks |
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7715 * of the buffer_size. This means we need roughly twice the memory |
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7716 * to load a file. |
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|
7717 */ |
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|
7718 ALint sound_duration = Sound_GetDuration(sample); |
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|
7719 if(sound_duration > 0) |
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|
7720 { |
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|
7721 size_t total_bytes = Compute_Total_Bytes_With_Frame_Padding(&sample->desired, (ALuint)sound_duration); |
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|
7722 int buffer_resize_succeeded = Sound_SetBufferSize(sample, total_bytes); |
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|
7723 if(0 == buffer_resize_succeeded) |
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|
7724 { |
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|
7725 ALmixer_SetError(Sound_GetError()); |
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|
7726 Sound_FreeSample(sample); |
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|
7727 free(ret_data); |
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|
7728 return NULL; |
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|
7729 } |
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|
7730 } |
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|
7731 #endif /* ALMIXER_DISABLE_PREDECODED_PRECOMPUTE_BUFFER_SIZE_OPTIMIZATION */ |
| 0 | 7732 bytes_decoded = Sound_DecodeAll(sample); |
| 7733 if(sample->flags & SOUND_SAMPLEFLAG_ERROR) | |
| 7734 { | |
| 7735 ALmixer_SetError(Sound_GetError()); | |
| 7736 Sound_FreeSample(sample); | |
| 7737 free(ret_data); | |
| 7738 return NULL; | |
| 7739 } | |
| 7740 | |
| 7741 /* If no data, return an error */ | |
| 7742 if(0 == bytes_decoded) | |
| 7743 { | |
| 7744 ALmixer_SetError("File has no data"); | |
| 7745 Sound_FreeSample(sample); | |
| 7746 free(ret_data); | |
| 7747 return NULL; | |
| 7748 } | |
| 7749 | |
| 7750 | |
| 7751 ret_data->decoded_all = 1; | |
| 7752 /* Need to keep this information around for | |
| 7753 * seek and rewind abilities. | |
| 7754 */ | |
| 7755 ret_data->total_bytes = bytes_decoded; | |
| 7756 /* For now, the loaded bytes is the same as total bytes, but | |
| 7757 * this could change during a seek operation | |
| 7758 */ | |
| 7759 ret_data->loaded_bytes = bytes_decoded; | |
| 7760 | |
| 7761 /* Let's compute the total playing time | |
| 7762 * SDL_sound does not yet provide this (we're working on | |
| 7763 * that at the moment...) | |
| 7764 */ | |
| 7765 ret_data->total_time = Compute_Total_Time(&sample->desired, bytes_decoded); | |
| 7766 | |
| 7767 /* Create one element in the buffer array for data for OpanAL */ | |
| 7768 ret_data->buffer = (ALuint*)malloc( sizeof(ALuint) ); | |
| 7769 if(NULL == ret_data->buffer) | |
| 7770 { | |
| 7771 ALmixer_SetError("Out of Memory"); | |
| 7772 Sound_FreeSample(sample); | |
| 7773 free(ret_data); | |
| 7774 return NULL; | |
| 7775 } | |
| 7776 /* Clear the error code */ | |
| 7777 alGetError(); | |
| 7778 /* Now generate an OpenAL buffer using that first element */ | |
| 7779 alGenBuffers(1, ret_data->buffer); | |
| 7780 if( (error = alGetError()) != AL_NO_ERROR) | |
| 7781 { | |
|
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|
7782 ALmixer_SetError("alGenBuffers failed: %s\n", alGetString(error)); |
| 0 | 7783 Sound_FreeSample(sample); |
| 7784 free(ret_data->buffer); | |
| 7785 free(ret_data); | |
| 7786 return NULL; | |
| 7787 } | |
|
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|
7788 /* |
| 0 | 7789 fprintf(stderr, "Actual rate=%d, desired=%d\n", sample->actual.rate, sample->desired.rate); |
|
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|
7790 */ |
| 0 | 7791 /* Now copy the data to the OpenAL buffer */ |
| 7792 /* We can't just set a pointer because the API needs | |
| 7793 * its own copy to assist hardware acceleration */ | |
| 7794 alBufferData(ret_data->buffer[0], | |
| 7795 TranslateFormat(&sample->desired), | |
| 7796 sample->buffer, | |
| 7797 bytes_decoded, | |
| 7798 sample->desired.rate | |
| 7799 ); | |
| 7800 if( (error = alGetError()) != AL_NO_ERROR) | |
| 7801 { | |
|
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|
7802 ALmixer_SetError("alBufferData failed: %s\n", alGetString(error)); |
| 0 | 7803 Sound_FreeSample(sample); |
| 7804 alDeleteBuffers(1, ret_data->buffer); | |
| 7805 free(ret_data->buffer); | |
| 7806 free(ret_data); | |
| 7807 return NULL; | |
| 7808 } | |
| 7809 | |
| 7810 /* We should be done with the sample since it's all | |
| 7811 * predecoded. So we can free the memory */ | |
| 7812 /* Need to keep around because Seek() needs it */ | |
| 7813 | |
| 7814 /* Additional notes: | |
| 7815 * We need to keep data around in case Seek() is needed | |
| 7816 * or other Sound_AudioInfo is needed. | |
| 7817 * This can either be done by not deleting the sample, | |
| 7818 * or it can be done by dynamically recreating it | |
| 7819 * when we need it. | |
| 7820 * Update: I think now it's up to the user by passing the | |
| 7821 * access_data flag. If they set the flag, then they get | |
| 7822 * data callbacks and seek support. If not, then they can | |
| 7823 * get all that stuff at the expense of keeping extra memory | |
| 7824 * around. | |
| 7825 */ | |
| 7826 if(0 == access_data) | |
| 7827 { | |
| 7828 Sound_FreeSample(sample); | |
| 7829 ret_data->sample = NULL; | |
| 7830 } | |
| 7831 | |
| 7832 /* Else, We keep a copy of the sample around. | |
| 7833 * so don't do anything. | |
| 7834 */ | |
| 7835 #if 0 | |
| 7836 #if defined(DISABLE_PREDECODED_SEEK) | |
| 7837 Sound_FreeSample(sample); | |
| 7838 ret_data->sample = NULL; | |
| 7839 #elif !defined(DISABLE_SEEK_MEMORY_OPTIMIZATION) | |
| 7840 Sound_FreeSample(sample); | |
| 7841 ret_data->sample = NULL; | |
| 7842 #else | |
| 7843 /* We keep a copy of the sample around. | |
| 7844 * so don't do anything. | |
| 7845 */ | |
| 7846 #endif | |
| 7847 #endif | |
| 7848 | |
| 7849 /* okay we're done here */ | |
| 7850 } | |
| 7851 else | |
| 7852 { | |
| 7853 /* Shouldn't get here */ | |
| 7854 ALmixer_SetError("Unknown decode mode"); | |
| 7855 Sound_FreeSample(sample); | |
| 7856 free(ret_data); | |
| 7857 return NULL; | |
| 7858 } | |
| 7859 | |
| 7860 return ret_data; | |
| 7861 } | |
| 7862 | |
| 7863 | |
| 7864 /* This will load a sample for us. Most of the uglyness is | |
| 7865 * error checking and the fact that streamed/predecoded files | |
| 7866 * must be treated differently. | |
| 7867 * I don't like the AudioInfo parameter. I removed it once, | |
| 7868 * but the system will fail on RAW samples because the user | |
| 7869 * must specify it, so I had to bring it back. | |
| 7870 * Remember I must close the rwops if there is an error before NewSample() | |
| 7871 */ | |
|
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7872 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 | 7873 { |
| 7874 Sound_Sample* sample = NULL; | |
| 7875 Sound_AudioInfo target; | |
| 7876 | |
| 7877 /* Initialize target values to defaults | |
| 7878 * 0 tells SDL_sound to use the "actual" values | |
| 7879 */ | |
| 7880 target.channels = 0; | |
| 7881 target.rate = 0; | |
| 7882 #if 0 | |
| 7883 /* This requires my new additions to SDL_sound. It will | |
| 7884 * convert the sample to the proper endian order. | |
| 7885 * If the actual is 8-bit, it will do unsigned, if | |
| 7886 * the actual is 16-bit, it will do signed. | |
| 7887 * I'm told by Ryan Gordon that OpenAL prefers the signedness | |
| 7888 * in this way. | |
| 7889 */ | |
| 7890 target.format = AUDIO_U8S16SYS; | |
| 7891 #else | |
| 7892 target.format = AUDIO_S16SYS; | |
| 7893 #endif | |
| 7894 | |
| 7895 /* Set a default buffersize if needed */ | |
| 7896 if(0 == buffersize) | |
| 7897 { | |
| 7898 buffersize = ALMIXER_DEFAULT_BUFFERSIZE; | |
| 7899 } | |
| 7900 | |
| 7901 sample = Sound_NewSample(rwops, fileext, &target, buffersize); | |
| 7902 if(NULL == sample) | |
| 7903 { | |
| 7904 ALmixer_SetError(Sound_GetError()); | |
| 7905 return NULL; | |
| 7906 } | |
| 7907 | |
| 1 | 7908 return( DoLoad(sample, buffersize, decode_mode_is_predecoded, max_queue_buffers, num_startup_buffers, access_data)); |
| 0 | 7909 } |
| 7910 | |
| 7911 | |
| 7912 | |
| 7913 /* This will load a sample for us from | |
| 7914 * a file (instead of RWops). Most of the uglyness is | |
| 7915 * error checking and the fact that streamed/predecoded files | |
| 7916 * must be treated differently. | |
| 7917 */ | |
|
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|
7918 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 | 7919 { |
| 7920 Sound_Sample* sample = NULL; | |
| 7921 Sound_AudioInfo target; | |
| 7922 | |
| 7923 /* Initialize target values to defaults | |
| 7924 * 0 tells SDL_sound to use the "actual" values | |
| 7925 */ | |
| 7926 target.channels = 0; | |
| 7927 target.rate = 0; | |
| 7928 | |
| 7929 #if 0 | |
| 7930 /* This requires my new additions to SDL_sound. It will | |
| 7931 * convert the sample to the proper endian order. | |
| 7932 * If the actual is 8-bit, it will do unsigned, if | |
| 7933 * the actual is 16-bit, it will do signed. | |
| 7934 * I'm told by Ryan Gordon that OpenAL prefers the signedness | |
| 7935 * in this way. | |
| 7936 */ | |
| 7937 target.format = AUDIO_U8S16SYS; | |
| 7938 #else | |
| 7939 target.format = AUDIO_S16SYS; | |
| 7940 #endif | |
| 7941 | |
| 7942 #if 0 | |
| 7943 /* Okay, here's a messy hack. The problem is that we need | |
| 7944 * to convert the sample to have the correct bitdepth, | |
| 7945 * endian order, and signedness values. | |
| 7946 * The bit depth is 8 or 16. | |
| 7947 * The endian order is the native order of the system. | |
| 7948 * The signedness depends on what the original value | |
| 7949 * of the sample. Unfortunately, we can't specify these | |
| 7950 * values until we after we already know what the original | |
| 7951 * values were for bitdepth and signedness. | |
| 7952 * So we must open the file once to get the values, | |
| 7953 * then close it, and then reopen it with the | |
| 7954 * correct desired target values. | |
| 7955 * I tried changing the sample->desired field after | |
| 7956 * the NewSample call, but it had no effect, so | |
| 7957 * it looks like it must be set on open. | |
| 7958 */ | |
| 7959 /* Pick a small buffersize for the first open to not | |
| 7960 * waste much time allocating memory */ | |
| 7961 sample = Sound_NewSampleFromFile(filename, NULL, 512); | |
| 7962 if(NULL == sample) | |
| 7963 { | |
| 7964 ALmixer_SetError(Sound_GetError()); | |
| 7965 return NULL; | |
| 7966 } | |
| 7967 | |
| 7968 bit_depth = GetBitDepth(sample->actual.format); | |
| 7969 signedness_value = GetSignednessValue(sample->actual.format); | |
| 7970 if(8 == bit_depth) | |
| 7971 { | |
| 7972 /* If 8 bit, then we don't have to worry about | |
| 7973 * endian issues. We can just use the actual format | |
| 7974 * value and it should do the right thing | |
| 7975 */ | |
| 7976 target.format = sample->actual.format; | |
| 7977 } | |
| 7978 else | |
| 7979 { | |
| 7980 /* We'll assume it's 16-bit, and if it's not | |
| 7981 * hopefully SDL_sound will return an error, | |
| 7982 * or let us convert to 16-bit | |
| 7983 */ | |
| 7984 /* Now we need to get the correct signedness */ | |
| 7985 if(ALMIXER_UNSIGNED_VALUE == signedness_value) | |
| 7986 { | |
| 7987 /* Set to Unsigned 16-bit, system endian order */ | |
| 7988 target.format = AUDIO_U16SYS; | |
| 7989 } | |
| 7990 else | |
| 7991 { | |
| 7992 /* Again, we'll assume it's Signed 16-bit system order | |
| 7993 * or force the conversion and hope it works out | |
| 7994 */ | |
| 7995 target.format = AUDIO_S16SYS; | |
| 7996 } | |
| 7997 } | |
| 7998 | |
| 7999 /* Now we have the correct info. We need to close and reopen */ | |
| 8000 Sound_FreeSample(sample); | |
| 8001 #endif | |
| 8002 | |
| 8003 sample = Sound_NewSampleFromFile(filename, &target, buffersize); | |
| 8004 if(NULL == sample) | |
| 8005 { | |
| 8006 ALmixer_SetError(Sound_GetError()); | |
| 8007 return NULL; | |
| 8008 } | |
| 8009 | |
|
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|
8010 /* |
| 0 | 8011 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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diff
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|
8012 */ |
| 1 | 8013 return( DoLoad(sample, buffersize, decode_mode_is_predecoded, max_queue_buffers, num_startup_buffers, access_data)); |
| 0 | 8014 } |
| 8015 | |
| 8016 | |
| 8017 /* This is a back door for RAW samples or if you need the | |
| 8018 * AudioInfo field. Use at your own risk. | |
| 8019 */ | |
|
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|
8020 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 | 8021 { |
| 8022 Sound_Sample* sample = NULL; | |
| 1 | 8023 Sound_AudioInfo sound_desired; |
| 8024 /* Rather than copying the data from struct to struct, I could just | |
| 8025 * cast the thing since the structs are meant to be identical. | |
| 8026 * But if SDL_sound changes it's implementation, bad things | |
| 8027 * will probably happen. (Or if I change my implementation and | |
| 8028 * forget about the cast, same bad scenario.) Since this is a load | |
| 8029 * function, performance of this is negligible. | |
| 8030 */ | |
| 8031 if(NULL == desired) | |
| 8032 { | |
| 8033 sample = Sound_NewSample(rwops, fileext, NULL, buffersize); | |
| 8034 } | |
| 8035 else | |
| 8036 { | |
| 8037 sound_desired.format = desired->format; | |
| 8038 sound_desired.channels = desired->channels; | |
| 8039 sound_desired.rate = desired->rate; | |
| 8040 sample = Sound_NewSample(rwops, fileext, &sound_desired, buffersize); | |
| 8041 } | |
| 0 | 8042 if(NULL == sample) |
| 8043 { | |
| 8044 ALmixer_SetError(Sound_GetError()); | |
| 8045 return NULL; | |
| 8046 } | |
| 1 | 8047 return( DoLoad(sample, buffersize, decode_mode_is_predecoded, max_queue_buffers, num_startup_buffers, access_data)); |
| 0 | 8048 } |
| 8049 | |
| 8050 | |
| 8051 | |
| 8052 | |
| 8053 /* This is a back door for RAW samples or if you need the | |
| 8054 * AudioInfo field. Use at your own risk. | |
| 8055 */ | |
|
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|
8056 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 | 8057 { |
| 8058 Sound_Sample* sample = NULL; | |
| 1 | 8059 Sound_AudioInfo sound_desired; |
| 8060 /* Rather than copying the data from struct to struct, I could just | |
| 8061 * cast the thing since the structs are meant to be identical. | |
| 8062 * But if SDL_sound changes it's implementation, bad things | |
| 8063 * will probably happen. (Or if I change my implementation and | |
| 8064 * forget about the cast, same bad scenario.) Since this is a load | |
| 8065 * function, performance of this is negligible. | |
| 8066 */ | |
| 8067 if(NULL == desired) | |
| 8068 { | |
| 8069 sample = Sound_NewSampleFromFile(filename, NULL, buffersize); | |
| 8070 } | |
| 8071 else | |
| 8072 { | |
| 8073 sound_desired.format = desired->format; | |
| 8074 sound_desired.channels = desired->channels; | |
| 8075 sound_desired.rate = desired->rate; | |
| 8076 sample = Sound_NewSampleFromFile(filename, &sound_desired, buffersize); | |
| 8077 } | |
| 8078 | |
| 0 | 8079 if(NULL == sample) |
| 8080 { | |
| 8081 ALmixer_SetError(Sound_GetError()); | |
| 8082 return NULL; | |
| 8083 } | |
| 1 | 8084 return( DoLoad(sample, buffersize, decode_mode_is_predecoded, max_queue_buffers, num_startup_buffers, access_data)); |
| 0 | 8085 } |
| 8086 | |
| 8087 | |
| 8088 | |
| 8089 | |
| 8090 void ALmixer_FreeData(ALmixer_Data* data) | |
| 8091 { | |
| 8092 ALenum error; | |
| 8093 if(NULL == data) | |
| 8094 { | |
| 8095 return; | |
| 8096 } | |
| 8097 | |
| 8098 if(data->decoded_all) | |
| 8099 { | |
| 8100 /* If access_data was enabled, then the Sound_Sample* | |
| 8101 * still exists. We need to free it | |
| 8102 */ | |
| 8103 if(data->sample != NULL) | |
| 8104 { | |
| 8105 Sound_FreeSample(data->sample); | |
| 8106 } | |
| 8107 alDeleteBuffers(1, data->buffer); | |
| 8108 if((error = alGetError()) != AL_NO_ERROR) | |
| 8109 { | |
| 8110 fprintf(stderr, "70Testing error: %s\n", | |
|
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|
8111 alGetString(error)); |
| 0 | 8112 } |
| 8113 | |
| 8114 } | |
| 8115 else | |
| 8116 { | |
|
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|
8117 ALuint i; |
| 0 | 8118 |
| 8119 /* Delete buffer copies if access_data was enabled */ | |
| 8120 if(data->buffer_map_list != NULL) | |
| 8121 { | |
| 8122 for(i=0; i<data->max_queue_buffers; i++) | |
| 8123 { | |
| 8124 free(data->buffer_map_list[i].data); | |
| 8125 } | |
| 8126 free(data->buffer_map_list); | |
| 8127 } | |
| 8128 if(data->circular_buffer_queue != NULL) | |
| 8129 { | |
| 8130 CircularQueueUnsignedInt_FreeQueue(data->circular_buffer_queue); | |
| 8131 } | |
| 8132 | |
| 8133 Sound_FreeSample(data->sample); | |
| 8134 alDeleteBuffers(data->max_queue_buffers, data->buffer); | |
| 8135 if((error = alGetError()) != AL_NO_ERROR) | |
| 8136 { | |
| 8137 fprintf(stderr, "71Testing error: %s\n", | |
|
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|
8138 alGetString(error)); |
| 0 | 8139 } |
| 8140 } | |
| 8141 free(data->buffer); | |
| 8142 free(data); | |
| 8143 } | |
| 8144 | |
|
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8145 ALint ALmixer_GetTotalTime(ALmixer_Data* data) |
|
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|
8146 { |
|
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|
8147 if(NULL == data) |
|
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|
8148 { |
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|
8149 return -1; |
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|
8150 } |
| 0 | 8151 return data->total_time; |
| 8152 } | |
| 8153 | |
| 8154 /* This function will look up the source for the corresponding channel */ | |
| 8155 /* Must return 0 on error instead of -1 because of unsigned int */ | |
|
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|
8156 ALuint ALmixer_GetSource(ALint channel) |
| 0 | 8157 { |
| 8158 ALuint retval; | |
|
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8159 #ifdef ENABLE_ALMIXER_THREADS |
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|
8160 SDL_LockMutex(s_simpleLock); |
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|
8161 #endif |
| 0 | 8162 retval = Internal_GetSource(channel); |
|
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8163 #ifdef ENABLE_ALMIXER_THREADS |
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|
8164 SDL_UnlockMutex(s_simpleLock); |
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|
8165 #endif |
| 0 | 8166 return retval; |
| 8167 } | |
| 8168 | |
| 8169 /* This function will look up the channel for the corresponding source */ | |
|
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|
8170 ALint ALmixer_GetChannel(ALuint source) |
|
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|
8171 { |
|
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|
8172 ALint retval; |
|
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|
8173 #ifdef ENABLE_ALMIXER_THREADS |
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|
8174 SDL_LockMutex(s_simpleLock); |
|
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|
8175 #endif |
| 0 | 8176 retval = Internal_GetChannel(source); |
|
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|
8177 #ifdef ENABLE_ALMIXER_THREADS |
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|
8178 SDL_UnlockMutex(s_simpleLock); |
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|
8179 #endif |
| 0 | 8180 return retval; |
| 8181 } | |
| 8182 | |
|
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|
8183 ALint ALmixer_FindFreeChannel(ALint start_channel) |
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8184 { |
|
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|
8185 ALint retval; |
|
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|
8186 #ifdef ENABLE_ALMIXER_THREADS |
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|
8187 SDL_LockMutex(s_simpleLock); |
|
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|
8188 #endif |
| 0 | 8189 retval = Internal_FindFreeChannel(start_channel); |
|
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|
8190 #ifdef ENABLE_ALMIXER_THREADS |
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|
8191 SDL_UnlockMutex(s_simpleLock); |
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|
8192 #endif |
| 0 | 8193 return retval; |
| 8194 } | |
| 8195 | |
| 8196 | |
| 8197 | |
| 8198 /* API update function. | |
| 8199 * It should return the number of buffers that were | |
| 8200 * queued during the call. The value might be | |
| 8201 * used to guage how long you might wait to | |
| 8202 * call the next update loop in case you are worried | |
| 8203 * about preserving CPU cycles. The idea is that | |
| 8204 * when a buffer is queued, there was probably some | |
| 8205 * CPU intensive looping which took awhile. | |
| 8206 * It's mainly provided as a convenience. | |
|
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|
8207 * Timing the call with ALmixer_GetTicks() would produce |
| 0 | 8208 * more accurate information. |
| 8209 * Returns a negative value if there was an error, | |
| 8210 * the value being the number of errors. | |
| 8211 */ | |
|
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|
8212 ALint ALmixer_Update() |
| 0 | 8213 { |
| 8214 #ifdef ENABLE_ALMIXER_THREADS | |
| 8215 /* The thread will handle all updates by itself. | |
| 8216 * Don't allow the user to explicitly call update. | |
| 8217 */ | |
| 8218 return 0; | |
| 8219 #else | |
| 8220 return( Update_ALmixer(NULL) ); | |
| 8221 #endif | |
| 8222 } | |
| 8223 | |
| 8224 | |
| 8225 | |
|
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|
8226 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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|
8227 { |
|
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|
8228 #ifdef ENABLE_ALMIXER_THREADS |
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|
8229 SDL_LockMutex(s_simpleLock); |
|
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|
8230 #endif |
|
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|
8231 Channel_Done_Callback = playback_finished_callback; |
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|
8232 Channel_Done_Callback_Userdata = user_data; |
|
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|
8233 #ifdef ENABLE_ALMIXER_THREADS |
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|
8234 SDL_UnlockMutex(s_simpleLock); |
|
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|
8235 #endif |
|
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|
8236 } |
|
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|
8237 |
|
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|
8238 |
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|
8239 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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|
8240 { |
|
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|
8241 #ifdef ENABLE_ALMIXER_THREADS |
|
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|
8242 SDL_LockMutex(s_simpleLock); |
|
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|
8243 #endif |
|
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|
8244 Channel_Data_Callback = playback_data_callback; |
| 1 | 8245 Channel_Data_Callback_Userdata = user_data; |
|
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|
8246 #ifdef ENABLE_ALMIXER_THREADS |
|
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|
8247 SDL_UnlockMutex(s_simpleLock); |
|
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|
8248 #endif |
|
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|
8249 } |
|
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|
8250 |
|
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|
8251 |
|
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|
8252 |
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|
8253 |
|
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|
8254 |
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|
8255 ALint ALmixer_PlayChannelTimed(ALint channel, ALmixer_Data* data, ALint loops, ALint ticks) |
|
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|
8256 { |
|
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|
8257 ALint retval; |
|
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|
8258 #ifdef ENABLE_ALMIXER_THREADS |
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|
8259 SDL_LockMutex(s_simpleLock); |
|
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|
8260 #endif |
| 0 | 8261 retval = Internal_PlayChannelTimed(channel, data, loops, ticks); |
|
2
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|
8262 #ifdef ENABLE_ALMIXER_THREADS |
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|
8263 SDL_UnlockMutex(s_simpleLock); |
|
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|
8264 #endif |
| 0 | 8265 return retval; |
| 8266 } | |
| 8267 | |
| 8268 | |
| 8269 /* In case the user wants to specify a source instead of a channel, | |
| 8270 * they may use this function. This function will look up the | |
| 8271 * source-to-channel map, and convert the call into a | |
| 8272 * PlayChannelTimed() function call. | |
| 8273 * Returns the channel it's being played on. | |
| 8274 * Note: If you are prefer this method, then you need to be careful | |
| 8275 * about using PlayChannel, particularly if you request the | |
| 8276 * first available channels because source and channels have | |
| 8277 * a one-to-one mapping in this API. It is quite easy for | |
| 8278 * a channel/source to already be in use because of this. | |
| 8279 * In this event, an error message will be returned to you. | |
| 8280 */ | |
|
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|
8281 ALuint ALmixer_PlaySourceTimed(ALuint source, ALmixer_Data* data, ALint loops, ALint ticks) |
| 0 | 8282 { |
| 8283 ALuint retval; | |
|
2
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|
8284 #ifdef ENABLE_ALMIXER_THREADS |
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|
8285 SDL_LockMutex(s_simpleLock); |
|
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|
8286 #endif |
| 0 | 8287 retval = Internal_PlaySourceTimed(source, data, loops, ticks); |
|
2
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|
8288 #ifdef ENABLE_ALMIXER_THREADS |
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|
8289 SDL_UnlockMutex(s_simpleLock); |
|
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|
8290 #endif |
| 0 | 8291 return retval; |
| 8292 } | |
| 8293 | |
| 8294 | |
| 8295 /* Will return the number of channels halted | |
| 8296 * or 0 for error | |
| 8297 */ | |
|
2
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|
8298 ALint ALmixer_HaltChannel(ALint channel) |
|
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|
8299 { |
|
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|
8300 ALint retval; |
|
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|
8301 #ifdef ENABLE_ALMIXER_THREADS |
|
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|
8302 SDL_LockMutex(s_simpleLock); |
|
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|
8303 #endif |
|
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|
8304 retval = Internal_HaltChannel(channel, AL_FALSE); |
|
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|
8305 #ifdef ENABLE_ALMIXER_THREADS |
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|
8306 SDL_UnlockMutex(s_simpleLock); |
|
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|
8307 #endif |
| 0 | 8308 return retval; |
| 8309 } | |
| 8310 | |
| 8311 /* Will return the number of channels halted | |
| 8312 * or 0 for error | |
| 8313 */ | |
|
2
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|
8314 ALint ALmixer_HaltSource(ALuint source) |
|
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|
8315 { |
|
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|
8316 ALint retval; |
|
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|
8317 #ifdef ENABLE_ALMIXER_THREADS |
|
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|
8318 SDL_LockMutex(s_simpleLock); |
|
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|
8319 #endif |
|
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|
8320 retval = Internal_HaltSource(source, AL_FALSE); |
|
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|
8321 #ifdef ENABLE_ALMIXER_THREADS |
|
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|
8322 SDL_UnlockMutex(s_simpleLock); |
|
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|
8323 #endif |
| 0 | 8324 return retval; |
| 8325 } | |
| 8326 | |
| 8327 | |
| 8328 /* This will rewind the SDL_Sound sample for streamed | |
| 8329 * samples and start buffering up the data for the next | |
| 8330 * playback. This may require samples to be halted | |
| 8331 */ | |
|
12
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|
8332 ALboolean ALmixer_RewindData(ALmixer_Data* data) |
|
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|
8333 { |
|
bfe90b4f3d87
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|
8334 ALboolean retval; |
|
2
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|
8335 #ifdef ENABLE_ALMIXER_THREADS |
|
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|
8336 SDL_LockMutex(s_simpleLock); |
|
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|
8337 #endif |
| 0 | 8338 retval = Internal_RewindData(data); |
|
2
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|
8339 #ifdef ENABLE_ALMIXER_THREADS |
|
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|
8340 SDL_UnlockMutex(s_simpleLock); |
|
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|
8341 #endif |
| 0 | 8342 return retval; |
| 8343 } | |
| 8344 | |
|
2
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|
8345 ALint ALmixer_RewindChannel(ALint channel) |
|
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|
8346 { |
|
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|
8347 ALint retval; |
|
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|
8348 #ifdef ENABLE_ALMIXER_THREADS |
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|
8349 SDL_LockMutex(s_simpleLock); |
|
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|
8350 #endif |
| 0 | 8351 retval = Internal_RewindChannel(channel); |
|
2
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|
8352 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8353 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8354 #endif |
| 0 | 8355 return retval; |
| 8356 } | |
| 8357 | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8358 ALint ALmixer_RewindSource(ALuint source) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8359 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8360 ALint retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8361 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8362 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8363 #endif |
| 0 | 8364 retval = Internal_RewindSource(source); |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8365 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8366 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8367 #endif |
| 0 | 8368 return retval; |
| 8369 } | |
| 8370 | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8371 ALint ALmixer_PauseChannel(ALint channel) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8372 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8373 ALint retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8374 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8375 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8376 #endif |
| 0 | 8377 retval = Internal_PauseChannel(channel); |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8378 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8379 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8380 #endif |
| 0 | 8381 return retval; |
| 8382 } | |
| 8383 | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8384 ALint ALmixer_PauseSource(ALuint source) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8385 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8386 ALint retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8387 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8388 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8389 #endif |
| 0 | 8390 retval = Internal_PauseSource(source); |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8391 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8392 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8393 #endif |
| 0 | 8394 return retval; |
| 8395 } | |
| 8396 | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8397 ALint ALmixer_ResumeChannel(ALint channel) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8398 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8399 ALint retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8400 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8401 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8402 #endif |
| 0 | 8403 retval = Internal_ResumeChannel(channel); |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8404 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8405 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8406 #endif |
| 0 | 8407 return retval; |
| 8408 } | |
| 8409 | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8410 ALint ALmixer_ResumeSource(ALuint source) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8411 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8412 ALint retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8413 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8414 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8415 #endif |
| 0 | 8416 retval = Internal_ResumeSource(source); |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8417 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8418 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8419 #endif |
| 0 | 8420 return retval; |
| 8421 } | |
| 8422 | |
| 8423 /* Might consider setting eof to 0 as a "feature" | |
| 8424 * This will allow seek to end to stay there because | |
| 8425 * Play automatically rewinds if at the end */ | |
|
12
bfe90b4f3d87
Bug fixes to FadeIn.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
10
diff
changeset
|
8426 ALboolean ALmixer_SeekData(ALmixer_Data* data, ALuint msec) |
|
bfe90b4f3d87
Bug fixes to FadeIn.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
10
diff
changeset
|
8427 { |
|
bfe90b4f3d87
Bug fixes to FadeIn.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
10
diff
changeset
|
8428 ALboolean retval; |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8429 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8430 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8431 #endif |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8432 retval = Internal_SeekData(data, msec); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8433 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8434 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8435 #endif |
| 0 | 8436 return retval; |
| 8437 } | |
| 8438 | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8439 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
|
8440 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8441 ALint retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8442 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8443 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8444 #endif |
| 0 | 8445 retval = Internal_FadeInChannelTimed(channel, 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
|
8446 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8447 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8448 #endif |
| 0 | 8449 return retval; |
| 8450 } | |
| 8451 | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8452 ALuint ALmixer_FadeInSourceTimed(ALuint source, ALmixer_Data* data, ALint loops, ALuint fade_ticks, ALint expire_ticks) |
| 0 | 8453 { |
| 8454 ALuint retval; | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8455 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8456 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8457 #endif |
| 0 | 8458 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
|
8459 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8460 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8461 #endif |
| 0 | 8462 return retval; |
| 8463 } | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8464 |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8465 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
|
8466 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8467 ALint retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8468 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8469 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8470 #endif |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8471 retval = Internal_FadeOutChannel(channel, ticks); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8472 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8473 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8474 #endif |
| 0 | 8475 return retval; |
| 8476 } | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8477 |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8478 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
|
8479 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8480 ALint retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8481 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8482 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8483 #endif |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8484 retval = Internal_FadeOutSource(source, ticks); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8485 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8486 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8487 #endif |
| 0 | 8488 return retval; |
| 8489 } | |
| 8490 | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8491 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
|
8492 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8493 ALint retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8494 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8495 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8496 #endif |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8497 retval = Internal_FadeChannel(channel, ticks, volume); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8498 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8499 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8500 #endif |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8501 return retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8502 } |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8503 |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8504 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
|
8505 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8506 ALint retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8507 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8508 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8509 #endif |
| 0 | 8510 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
|
8511 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8512 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8513 #endif |
| 0 | 8514 return retval; |
| 8515 } | |
| 8516 | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8517 |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8518 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
|
8519 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8520 ALboolean retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8521 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8522 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8523 #endif |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8524 retval = Internal_SetVolumeChannel(channel, volume); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8525 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8526 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8527 #endif |
| 0 | 8528 return retval; |
| 8529 } | |
| 8530 | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8531 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
|
8532 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8533 ALboolean retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8534 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8535 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8536 #endif |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8537 retval = Internal_SetVolumeSource(source, volume); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8538 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8539 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8540 #endif |
| 0 | 8541 return retval; |
| 8542 } | |
| 8543 | |
|
2
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8544 ALfloat ALmixer_GetVolumeChannel(ALint channel) |
| 0 | 8545 { |
| 8546 ALfloat retval; | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8547 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8548 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8549 #endif |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8550 retval = Internal_GetVolumeChannel(channel); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8551 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8552 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8553 #endif |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8554 return retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8555 } |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8556 |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8557 ALfloat ALmixer_GetVolumeSource(ALuint source) |
|
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 ALfloat retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8560 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8561 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8562 #endif |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8563 retval = Internal_GetVolumeSource(source); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8564 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8565 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8566 #endif |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8567 return retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8568 } |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8569 |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8570 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
|
8571 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8572 ALboolean retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8573 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8574 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8575 #endif |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8576 retval = Internal_SetMaxVolumeChannel(channel, volume); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8577 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8578 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8579 #endif |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8580 return retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8581 } |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8582 |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8583 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
|
8584 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8585 ALboolean retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8586 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8587 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8588 #endif |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8589 retval = Internal_SetMaxVolumeSource(source, volume); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8590 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8591 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8592 #endif |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8593 return retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8594 } |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8595 |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8596 ALfloat ALmixer_GetMaxVolumeChannel(ALint channel) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8597 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8598 ALfloat retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8599 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8600 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8601 #endif |
| 0 | 8602 retval = Internal_GetMaxVolumeChannel(channel); |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8603 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8604 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8605 #endif |
| 0 | 8606 return retval; |
| 8607 } | |
| 8608 | |
| 8609 ALfloat ALmixer_GetMaxVolumeSource(ALuint source) | |
| 8610 { | |
| 8611 ALfloat retval; | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8612 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8613 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8614 #endif |
| 0 | 8615 retval = Internal_GetMaxVolumeSource(source); |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8616 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8617 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8618 #endif |
| 0 | 8619 return retval; |
| 8620 } | |
| 8621 | |
| 8622 | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8623 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
|
8624 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8625 ALboolean retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8626 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8627 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8628 #endif |
| 0 | 8629 retval = Internal_SetMinVolumeChannel(channel, volume); |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8630 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8631 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8632 #endif |
| 0 | 8633 return retval; |
| 8634 } | |
| 8635 | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8636 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
|
8637 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8638 ALboolean retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8639 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8640 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8641 #endif |
| 0 | 8642 retval = Internal_SetMinVolumeSource(source, volume); |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8643 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8644 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8645 #endif |
| 0 | 8646 return retval; |
| 8647 } | |
| 8648 | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8649 ALfloat ALmixer_GetMinVolumeChannel(ALint channel) |
| 0 | 8650 { |
| 8651 ALfloat retval; | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8652 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8653 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8654 #endif |
| 0 | 8655 retval = Internal_GetMinVolumeChannel(channel); |
|
2
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_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8658 #endif |
| 0 | 8659 return retval; |
| 8660 } | |
| 8661 | |
| 8662 ALfloat ALmixer_GetMinVolumeSource(ALuint source) | |
| 8663 { | |
| 8664 ALfloat retval; | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8665 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8666 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8667 #endif |
| 0 | 8668 retval = Internal_GetMinVolumeSource(source); |
|
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_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8671 #endif |
| 0 | 8672 return retval; |
| 8673 } | |
| 8674 | |
| 8675 | |
| 8676 | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8677 ALboolean ALmixer_SetMasterVolume(ALfloat volume) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8678 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8679 ALboolean retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8680 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8681 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8682 #endif |
| 0 | 8683 retval = Internal_SetMasterVolume(volume); |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8684 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8685 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8686 #endif |
| 0 | 8687 return retval; |
| 8688 } | |
| 8689 | |
| 8690 ALfloat ALmixer_GetMasterVolume() | |
| 8691 { | |
| 8692 ALfloat retval; | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8693 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8694 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8695 #endif |
| 0 | 8696 retval = Internal_GetMasterVolume(); |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8697 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8698 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8699 #endif |
| 0 | 8700 return retval; |
| 8701 } | |
| 8702 | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8703 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
|
8704 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8705 ALint retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8706 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8707 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8708 #endif |
| 0 | 8709 retval = Internal_ExpireChannel(channel, ticks); |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8710 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8711 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8712 #endif |
| 0 | 8713 return retval; |
| 8714 } | |
| 8715 | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8716 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
|
8717 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8718 ALint retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8719 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8720 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8721 #endif |
| 0 | 8722 retval = Internal_ExpireSource(source, ticks); |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8723 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8724 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8725 #endif |
| 0 | 8726 return retval; |
| 8727 } | |
| 8728 | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8729 ALint ALmixer_IsActiveChannel(ALint channel) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8730 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8731 ALint retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8732 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8733 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8734 #endif |
| 0 | 8735 retval = Internal_QueryChannel(channel); |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8736 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8737 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8738 #endif |
| 0 | 8739 return retval; |
| 8740 } | |
| 8741 | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8742 ALint ALmixer_IsActiveSource(ALuint source) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8743 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8744 ALint retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8745 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8746 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8747 #endif |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8748 retval = Internal_QuerySource(source); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8749 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8750 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8751 #endif |
| 0 | 8752 return retval; |
| 8753 } | |
| 8754 | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8755 |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8756 ALint ALmixer_IsPlayingChannel(ALint channel) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8757 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8758 ALint retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8759 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8760 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8761 #endif |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8762 retval = Internal_PlayingChannel(channel); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8763 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8764 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8765 #endif |
| 0 | 8766 return retval; |
| 8767 } | |
| 8768 | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8769 ALint ALmixer_IsPlayingSource(ALuint source) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8770 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8771 ALint retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8772 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8773 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8774 #endif |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8775 retval = Internal_PlayingSource(source); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8776 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8777 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8778 #endif |
| 0 | 8779 return retval; |
| 8780 } | |
| 8781 | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8782 |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8783 ALint ALmixer_IsPausedChannel(ALint channel) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8784 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8785 ALint retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8786 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8787 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8788 #endif |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8789 retval = Internal_PausedChannel(channel); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8790 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8791 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8792 #endif |
| 0 | 8793 return retval; |
| 8794 } | |
| 8795 | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8796 ALint ALmixer_IsPausedSource(ALuint source) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8797 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8798 ALint retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8799 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8800 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8801 #endif |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8802 retval = Internal_PausedSource(source); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8803 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8804 SDL_UnlockMutex(s_simpleLock); |
|
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|
8805 #endif |
| 0 | 8806 return retval; |
| 8807 } | |
| 8808 | |
|
2
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8809 |
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diff
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|
8810 ALuint ALmixer_CountAllFreeChannels() |
|
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parents:
1
diff
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|
8811 { |
|
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1
diff
changeset
|
8812 ALuint retval; |
|
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1
diff
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|
8813 #ifdef ENABLE_ALMIXER_THREADS |
|
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diff
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|
8814 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
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parents:
1
diff
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|
8815 #endif |
|
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diff
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|
8816 retval = Internal_CountAllFreeChannels(); |
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parents:
1
diff
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|
8817 #ifdef ENABLE_ALMIXER_THREADS |
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1
diff
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|
8818 SDL_UnlockMutex(s_simpleLock); |
|
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1
diff
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|
8819 #endif |
| 0 | 8820 return retval; |
| 8821 } | |
| 8822 | |
|
2
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diff
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|
8823 ALuint ALmixer_CountUnreservedFreeChannels() |
|
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diff
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|
8824 { |
|
279d0427ef26
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1
diff
changeset
|
8825 ALuint retval; |
|
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1
diff
changeset
|
8826 #ifdef ENABLE_ALMIXER_THREADS |
|
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1
diff
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|
8827 SDL_LockMutex(s_simpleLock); |
|
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1
diff
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|
8828 #endif |
|
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1
diff
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|
8829 retval = Internal_CountUnreservedFreeChannels(); |
|
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1
diff
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|
8830 #ifdef ENABLE_ALMIXER_THREADS |
|
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1
diff
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|
8831 SDL_UnlockMutex(s_simpleLock); |
|
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1
diff
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|
8832 #endif |
|
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diff
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|
8833 return retval; |
|
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diff
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|
8834 } |
|
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diff
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|
8835 |
|
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1
diff
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|
8836 ALuint ALmixer_CountAllUsedChannels() |
|
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1
diff
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|
8837 { |
|
279d0427ef26
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diff
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|
8838 ALuint retval; |
|
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diff
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|
8839 #ifdef ENABLE_ALMIXER_THREADS |
|
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diff
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|
8840 SDL_LockMutex(s_simpleLock); |
|
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1
diff
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|
8841 #endif |
| 0 | 8842 retval = Internal_CountAllUsedChannels(); |
|
2
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|
8843 #ifdef ENABLE_ALMIXER_THREADS |
|
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|
8844 SDL_UnlockMutex(s_simpleLock); |
|
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diff
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|
8845 #endif |
| 0 | 8846 return retval; |
| 8847 } | |
| 8848 | |
|
2
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|
8849 ALuint ALmixer_CountUnreservedUsedChannels() |
|
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1
diff
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|
8850 { |
|
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diff
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|
8851 ALuint retval; |
|
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diff
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|
8852 #ifdef ENABLE_ALMIXER_THREADS |
|
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diff
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|
8853 SDL_LockMutex(s_simpleLock); |
|
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diff
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|
8854 #endif |
| 0 | 8855 retval = Internal_CountUnreservedUsedChannels(); |
|
2
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diff
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|
8856 #ifdef ENABLE_ALMIXER_THREADS |
|
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diff
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|
8857 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
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diff
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|
8858 #endif |
| 0 | 8859 return retval; |
| 8860 } | |
| 8861 | |
|
2
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diff
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|
8862 ALboolean ALmixer_IsPredecoded(ALmixer_Data* data) |
| 1 | 8863 { |
| 8864 if(NULL == data) | |
| 8865 { | |
|
2
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diff
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|
8866 return AL_FALSE; |
| 1 | 8867 } |
| 8868 return data->decoded_all; | |
| 8869 } | |
| 8870 | |
|
2
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diff
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|
8871 ALboolean ALmixer_CompiledWithThreadBackend() |
|
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1
diff
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|
8872 { |
|
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1
diff
changeset
|
8873 #ifdef ENABLE_ALMIXER_THREADS |
|
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diff
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|
8874 return AL_TRUE; |
|
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diff
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|
8875 #else |
|
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diff
changeset
|
8876 return AL_FALSE; |
|
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diff
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|
8877 #endif |
|
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parents:
1
diff
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|
8878 } |
|
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Eric Wing <ewing . public |-at-| gmail . com>
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1
diff
changeset
|
8879 |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
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1
diff
changeset
|
8880 |
|
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1
diff
changeset
|
8881 |
|
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diff
changeset
|
8882 |