Mercurial > almixer_isolated
annotate ALmixer.c @ 3:a929285e1db0
Added CMake build system.
There are problems with the SDL_sound module due to changes in CMake. Right now they just seem to be warnings, but I am unable to suppress them.
Added license.
Added README.
author | Eric Wing <ewing . public |-at-| gmail . com> |
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date | Wed, 27 Oct 2010 20:43:14 -0700 |
parents | SDL_ALmixer.c@279d0427ef26 |
children | 4b1048af7e55 |
rev | line source |
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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 | |
3
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8 #include "ALmixer.h" |
0 | 9 |
2
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10 #ifdef ALMIXER_COMPILE_WITHOUT_SDL |
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11 #include "ALmixer_rwops.h" |
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12 #include "SoundDecoder.h" |
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13 #else |
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14 #include "SDL_sound.h" |
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15 #endif |
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16 |
0 | 17 #include "al.h" /* OpenAL */ |
18 #include "alc.h" /* For creating OpenAL contexts */ | |
19 | |
1 | 20 #ifdef __APPLE__ |
21 /* For performance things like ALC_CONVERT_DATA_UPON_LOADING */ | |
22 /* Note: ALC_CONVERT_DATA_UPON_LOADING used to be in the alc.h header. | |
23 * But in the Tiger OpenAL 1.1 release (10.4.7 and Xcode 2.4), the | |
24 * define was moved to a new header file and renamed to | |
25 * ALC_MAC_OSX_CONVERT_DATA_UPON_LOADING. | |
26 */ | |
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27 /* |
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28 #include <TargetConditionals.h> |
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29 #if (TARGET_OS_IPHONE == 1) || (TARGET_IPHONE_SIMULATOR == 1) |
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30 |
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31 #else |
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32 #include <OpenAL/MacOSX_OALExtensions.h> |
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33 #endif |
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34 */ |
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35 |
1 | 36 #endif |
37 | |
0 | 38 /* For malloc, bsearch, qsort */ |
39 #include <stdlib.h> | |
40 | |
41 /* For memcpy */ | |
42 #include <string.h> | |
43 | |
44 #if 0 | |
45 /* for toupper */ | |
46 #include <ctype.h> | |
47 /* for strrchr */ | |
48 #include <string.h> | |
49 #endif | |
50 | |
51 /* Currently used in the output debug functions */ | |
52 #include <stdio.h> | |
53 | |
54 /* My own CircularQueue implementation needed | |
55 * to work around the Nvidia problem of the | |
56 * lack of a buffer query. | |
57 */ | |
58 #include "CircularQueue.h" | |
59 | |
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60 #ifdef ENABLE_ALMIXER_THREADS |
0 | 61 /* Needed for the Mutex locks (and threads if enabled) */ |
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62 #ifdef ALMIXER_COMPILE_WITHOUT_SDL |
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63 #include "SimpleMutex.h" |
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64 #include "SimpleThread.h" |
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65 typedef struct SimpleMutex SDL_mutex; |
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66 typedef struct SimpleThread SDL_Thread; |
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67 #define SDL_CreateMutex SimpleMutex_CreateMutex |
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68 #define SDL_DestroyMutex SimpleMutex_DestroyMutex |
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69 #define SDL_LockMutex SimpleMutex_LockMutex |
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70 #define SDL_UnlockMutex SimpleMutex_UnlockMutex |
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71 #define SDL_CreateThread SimpleThread_CreateThread |
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72 #define SDL_WaitThread SimpleThread_WaitThread |
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73 |
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74 #else |
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75 #include "SDL_thread.h" |
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76 #endif |
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77 #endif |
0 | 78 |
79 /* Because of the API differences between the Loki | |
80 * and Creative distributions, we need to know which | |
81 * version to use. The LOKI distribution currently | |
82 * has AL_BYTE_LOKI defined in altypes.h which | |
83 * I will use as a flag to identify the distributions. | |
84 * If this is ever removed, I might revert back to the | |
85 * if defined(_WIN32) or defined(__APPLE__) test to | |
86 * identify the Creative dist. | |
87 * I'm not sure if or how the Nvidia distribution differs | |
88 * from the Creative distribution. So for | |
89 * now, the Nvidia distribution gets lumped with the | |
90 * Creative dist and I hope nothing will break. | |
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91 * My alGetString may be the most vulnerable. |
0 | 92 */ |
93 #ifdef AL_BYTE_LOKI | |
94 #define USING_LOKI_AL_DIST | |
95 /* This is a short term fix to get around the | |
96 * queuing problem with non-power of two buffer sizes. | |
97 * Hopefully the maintainers will fix this before | |
98 * we're ready to ship. | |
99 */ | |
100 #define ENABLE_LOKI_QUEUE_FIX_HACK | |
101 | |
102 /* The AL_GAIN in the Loki dist doesn't seem to do | |
103 * what I want/expect it to do. I want to use it for | |
104 * Fading, but it seems to work like an off/on switch. | |
105 * 0 = off, >0 = on. | |
106 * The AL_GAIN_LINEAR_LOKI switch seems to do what | |
107 * I want, so I'll redefine it here so the code is consistent | |
108 */ | |
109 /* Update: I've changed the source volume implementations | |
110 * to use AL_MAX_GAIN, so I don't think I need this block | |
111 * of code anymore. The listener uses AL_GAIN, but I | |
112 * hope they got this one right since there isn't a AL_MAX_GAIN | |
113 * for the listener. | |
114 */ | |
115 /* | |
116 #undef AL_GAIN | |
117 #include "alexttypes.h" | |
118 #define AL_GAIN AL_GAIN_LINEAR_LOKI | |
119 */ | |
120 #else | |
121 /* Might need to run other tests to figure out the DIST */ | |
122 /* I've been told that Nvidia doesn't define constants | |
123 * in the headers like Creative. Instead of | |
124 * #define AL_REFERENCE_DISTANCE 0x1020, | |
125 * Nvidia prefers you query OpenAL for a value. | |
126 * int AL_REFERENCE_DISTANCE = alGetEnumValue(ALubyte*)"AL_REFERNECE_DISTANCE"); | |
127 * So I'm assuming this means the Nvidia lacks this value. | |
128 * If this is the case, | |
129 * I guess we can use it to identify the Nvidia dist | |
130 */ | |
131 #ifdef AL_REFERENCE_DISTANCE | |
132 #define USING_CREATIVE_AL_DIST | |
133 #else | |
134 #define USING_NVIDIA_AL_DIST | |
135 #endif | |
136 #endif | |
137 | |
138 #ifdef ENABLE_LOKI_QUEUE_FIX_HACK | |
139 /* Need memset to zero out data */ | |
140 #include <string.h> | |
141 #endif | |
142 | |
143 | |
144 /* Seek issues for predecoded samples: | |
145 * The problem is that OpenAL makes us copy an | |
146 * entire buffer if we want to use it. This | |
147 * means we potentially have two copies of the | |
148 * same data. For predecoded data, this can be a | |
149 * large amount of memory. However, for seek | |
150 * support, I need to be able to get access to | |
151 * the original data so I can set byte positions. | |
152 * The following flags let you disable seek support | |
153 * if you don't want the memory hit, keep everything, | |
154 * or let you try to minimize the memory wasted by | |
155 * fetching it from the OpenAL buffer if needed | |
156 * and making a copy of it. | |
157 * Update: I don't think I need this flag anymore. I've made the | |
158 * effects of this user customizable by the access_data flag on load. | |
159 * If set to true, then seek and data callbacks work, with the | |
160 * cost of more memory and possibly CPU for copying the data through | |
161 * the callbacks. If false, then the extra memory is freed, but | |
162 * you don't get the features. | |
163 */ | |
164 /* | |
165 #define DISABLE_PREDECODED_SEEK | |
166 */ | |
167 /* Problem: Even though alGetBufferi(., AL_DATA, .) | |
168 * is in the Creative Programmer's reference, | |
169 * it actually isn't in the dist. (Invalid enum | |
170 * in Creative, can't compile in Loki.) | |
171 * So we have to keep it disabled | |
172 */ | |
173 #define DISABLE_SEEK_MEMORY_OPTIMIZATION | |
174 | |
175 #ifndef DISABLE_SEEK_MEMORY_OPTIMIZATION | |
176 /* Needed for memcpy */ | |
177 #include <string.h> | |
178 #endif | |
179 | |
180 /* Old way of doing things: | |
181 #if defined(_WIN32) || defined(__APPLE__) | |
182 #define USING_CREATIVE_AL_DIST | |
183 #else | |
184 #define USING_LOKI_AL_DIST | |
185 #endif | |
186 */ | |
187 | |
188 /************ REMOVE ME (Don't need anymore) ********/ | |
189 #if 0 | |
190 /* Let's get fancy and see if triple buffering | |
191 * does anything good for us | |
192 * Must be 2 or more or things will probably break | |
193 */ | |
194 #define NUMBER_OF_QUEUE_BUFFERS 5 | |
195 /* This is the number of buffers that are queued up | |
196 * when play first starts up. This should be at least 1 | |
197 * and no more than NUMBER_OF_QUEUE_BUFFERS | |
198 */ | |
199 #define NUMBER_OF_START_UP_BUFFERS 2 | |
200 #endif | |
201 /************ END REMOVE ME (Don't need anymore) ********/ | |
202 | |
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203 #ifdef ALMIXER_COMPILE_WITHOUT_SDL |
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204 #include "tErrorLib.h" |
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205 static TErrorPool* s_ALmixerErrorPool = NULL; |
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206 #endif |
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207 |
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208 static ALboolean ALmixer_Initialized = 0; |
0 | 209 /* This should be set correctly by Init */ |
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210 static ALuint ALmixer_Frequency_global = ALMIXER_DEFAULT_FREQUENCY; |
0 | 211 |
212 /* Will be initialized in Init */ | |
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213 static ALint Number_of_Channels_global = 0; |
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214 static ALint Number_of_Reserve_Channels_global = 0; |
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215 static ALuint Is_Playing_global = 0; |
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216 |
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217 #ifdef ENABLE_ALMIXER_THREADS |
0 | 218 /* This is for a simple lock system. It is not meant to be good, |
219 * but just sufficient to minimize/avoid threading issues | |
220 */ | |
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221 static SDL_mutex* s_simpleLock; |
0 | 222 static SDL_Thread* Stream_Thread_global = NULL; |
223 #endif | |
224 | |
225 | |
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226 #ifdef __APPLE__ |
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227 static ALvoid Internal_alcMacOSXMixerOutputRate(const ALdouble sample_rate) |
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228 { |
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229 static void (*alcMacOSXMixerOutputRateProcPtr)(const ALdouble) = NULL; |
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230 |
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231 if(NULL == alcMacOSXMixerOutputRateProcPtr) |
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232 { |
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233 alcMacOSXMixerOutputRateProcPtr = alGetProcAddress((const ALCchar*) "alcMacOSXMixerOutputRate"); |
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234 } |
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235 |
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236 if(NULL != alcMacOSXMixerOutputRateProcPtr) |
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237 { |
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238 alcMacOSXMixerOutputRateProcPtr(sample_rate); |
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239 } |
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240 |
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241 return; |
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242 } |
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243 |
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244 ALdouble Internal_alcMacOSXGetMixerOutputRate() |
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245 { |
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246 static ALdouble (*alcMacOSXGetMixerOutputRateProcPtr)(void) = NULL; |
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247 |
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248 if(NULL == alcMacOSXGetMixerOutputRateProcPtr) |
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249 { |
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250 alcMacOSXGetMixerOutputRateProcPtr = alGetProcAddress((const ALCchar*) "alcMacOSXGetMixerOutputRate"); |
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251 } |
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252 |
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253 if(NULL != alcMacOSXGetMixerOutputRateProcPtr) |
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254 { |
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255 return alcMacOSXGetMixerOutputRateProcPtr(); |
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256 } |
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257 |
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258 return 0.0; |
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259 } |
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260 #endif |
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261 |
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262 #ifdef ALMIXER_COMPILE_WITHOUT_SDL |
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263 |
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264 #if defined(__APPLE__) |
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265 #include <QuartzCore/QuartzCore.h> |
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266 #include <unistd.h> |
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267 static CFTimeInterval s_ticksBaseTime = 0.0; |
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268 |
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269 #elif defined(_WIN32) |
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270 #define WIN32_LEAN_AND_MEAN |
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271 #include <windows.h> |
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272 #include <winbase.h> |
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273 LARGE_INTEGER s_hiResTicksPerSecond; |
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274 double s_hiResSecondsPerTick; |
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275 LARGE_INTEGER s_ticksBaseTime; |
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276 #else |
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277 #include <unistd.h> |
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278 #include <time.h> |
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279 static struct timespec s_ticksBaseTime; |
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280 #endif |
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281 static void ALmixer_InitTime() |
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282 { |
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283 #if defined(__APPLE__) |
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284 s_ticksBaseTime = CACurrentMediaTime(); |
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285 |
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286 #elif defined(_WIN32) |
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287 LARGE_INTEGER hi_res_ticks_per_second; |
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288 if(TRUE == QueryPerformanceFrequency(&hi_res_ticks_per_second)) |
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289 { |
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290 QueryPerformanceCounter(&s_ticksBaseTime); |
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291 s_hiResSecondsPerTick = 1.0 / hi_res_ticks_per_second; |
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292 } |
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293 else |
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294 { |
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295 ALMixer_SetError("Windows error: High resolution clock failed."); |
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296 fprintf(stderr, "Windows error: High resolution clock failed. Audio will not work correctly.\n"); |
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297 } |
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298 #else |
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299 /* clock_gettime is POSIX.1-2001 */ |
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300 clock_gettime(CLOCK_MONOTONIC, &s_ticksBaseTime); |
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301 #endif |
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302 |
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303 } |
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304 static ALuint ALmixer_GetTicks() |
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305 { |
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306 #if defined(__APPLE__) |
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307 return (ALuint)((CACurrentMediaTime()-s_ticksBaseTime)*1000.0); |
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308 #elif defined(_WIN32) |
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309 LARGE_INTEGER current_time; |
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310 QueryPerformanceCounter(¤t_time); |
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311 return (ALuint)((current_time.QuadPart - s_ticksBaseTime.QuadPart) * 1000 * s_hiResSecondsPerTick); |
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312 |
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313 #else /* assuming POSIX */ |
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314 /* clock_gettime is POSIX.1-2001 */ |
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315 struct timespec current_time; |
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316 clock_gettime(CLOCK_MONOTONIC, ¤t_time); |
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317 return (ALuint)((current_time.tv_sec - s_ticksBaseTime.tv_sec)*1000.0 + (current_time.tv_nec - s_ticksBaseTime.tv_nsec) / 1000000); |
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318 #endif |
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319 } |
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320 static void ALmixer_Delay(ALuint milliseconds_delay) |
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321 { |
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322 #if defined(_WIN32) |
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323 Sleep(milliseconds_delay); |
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324 #else |
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325 usleep(milliseconds_delay); |
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326 #endif |
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327 } |
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328 #else |
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329 #include "SDL.h" /* For SDL_GetTicks(), SDL_Delay */ |
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330 #define ALmixer_GetTicks SDL_GetTicks |
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331 #define ALmixer_Delay SDL_Delay |
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332 #endif |
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333 |
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334 |
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335 |
0 | 336 /* If ENABLE_PARANOID_SIGNEDNESS_CHECK is used, |
337 * these values will be reset on Init() | |
338 * Consider these values Read-Only. | |
339 */ | |
340 | |
341 #define ALMIXER_SIGNED_VALUE 127 | |
342 #define ALMIXER_UNSIGNED_VALUE 255 | |
343 | |
344 #ifdef ENABLE_PARANOID_SIGNEDNESS_CHECK | |
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345 static ALushort SIGN_TYPE_16_BIT_FORMAT = AUDIO_S16SYS; |
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346 static ALushort SIGN_TYPE_8_BIT_FORMAT = AUDIO_S8; |
0 | 347 #else |
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348 static const ALushort SIGN_TYPE_16_BIT_FORMAT = AUDIO_S16SYS; |
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349 static const ALushort SIGN_TYPE_8_BIT_FORMAT = AUDIO_S8; |
0 | 350 #endif |
351 | |
1 | 352 |
353 /* This can be private instead of being in the header now that I moved | |
354 * ALmixer_Data inside here. | |
355 */ | |
356 typedef struct ALmixer_Buffer_Map ALmixer_Buffer_Map; | |
357 | |
358 | |
359 struct ALmixer_Data | |
360 { | |
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361 ALboolean decoded_all; /* dictates different behaviors */ |
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362 ALint total_time; /* total playing time of sample (msec) */ |
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363 |
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364 ALuint in_use; /* needed to prevent sharing for streams */ |
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365 ALboolean eof; /* flag for eof, only used for streams */ |
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366 |
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367 ALuint total_bytes; /* For predecoded */ |
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368 ALuint loaded_bytes; /* For predecoded (for seek) */ |
1 | 369 |
370 Sound_Sample* sample; /* SDL_Sound provides the data */ | |
371 ALuint* buffer; /* array of OpenAL buffers (at least 1 for predecoded) */ | |
372 | |
373 /* Needed for streamed buffers */ | |
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374 ALuint max_queue_buffers; /* Max number of queue buffers */ |
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375 ALuint num_startup_buffers; /* Number of ramp-up buffers */ |
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376 ALuint num_buffers_in_use; /* number of buffers in use */ |
1 | 377 |
378 /* This stuff is for streamed buffers that require data access */ | |
379 ALmixer_Buffer_Map* buffer_map_list; /* translate ALbuffer to index | |
380 and holds pointer to copy of data for | |
381 data access */ | |
382 ALuint current_buffer; /* The current playing buffer */ | |
383 | |
384 /* Nvidia distribution refuses to recognize a simple buffer query command | |
385 * unlike all other distributions. It's forcing me to redo the code | |
386 * to accomodate this Nvidia flaw by making me maintain a "best guess" | |
387 * copy of what I think the buffer queue state looks like. | |
388 * A circular queue would a helpful data structure for this task, | |
389 * but I wanted to avoid making an additional header requirement, | |
390 * so I'm making it a void* | |
391 */ | |
392 void* circular_buffer_queue; | |
393 | |
394 | |
395 }; | |
396 | |
0 | 397 static struct ALmixer_Channel |
398 { | |
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399 ALboolean channel_in_use; |
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400 ALboolean callback_update; /* For streaming determination */ |
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401 ALboolean needs_stream; /* For streaming determination */ |
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402 ALboolean halted; |
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403 ALboolean paused; |
0 | 404 ALuint alsource; |
405 ALmixer_Data* almixer_data; | |
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406 ALint loops; |
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407 ALint expire_ticks; |
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408 ALuint start_time; |
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409 |
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410 ALboolean fade_enabled; |
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411 ALuint fade_expire_ticks; |
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412 ALuint fade_start_time; |
0 | 413 ALfloat fade_inv_time; |
414 ALfloat fade_start_volume; | |
415 ALfloat fade_end_volume; | |
416 ALfloat max_volume; | |
417 ALfloat min_volume; | |
418 | |
419 /* Do we need other flags? | |
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420 ALbyte *samples; |
0 | 421 int volume; |
422 int looping; | |
423 int tag; | |
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424 ALuint expire; |
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425 ALuint start_time; |
0 | 426 Mix_Fading fading; |
427 int fade_volume; | |
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428 ALuint fade_length; |
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429 ALuint ticks_fade; |
0 | 430 effect_info *effects; |
431 */ | |
432 } *ALmixer_Channel_List = NULL; | |
433 | |
1 | 434 struct ALmixer_Buffer_Map |
435 { | |
436 ALuint albuffer; | |
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437 ALint index; /* might not need */ |
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438 ALbyte* data; |
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439 ALuint num_bytes; |
1 | 440 }; |
441 | |
0 | 442 /* This will be used to find a channel if the user supplies a source */ |
443 typedef struct Source_Map | |
444 { | |
445 ALuint source; | |
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446 ALint channel; |
0 | 447 } Source_Map; |
448 /* Keep an array of all sources with their associated channel */ | |
449 static Source_Map* Source_Map_List; | |
450 | |
451 static int Compare_Source_Map(const void* a, const void* b) | |
452 { | |
453 return ( ((Source_Map*)a)->source - ((Source_Map*)b)->source ); | |
454 } | |
455 | |
456 /* Sort by channel instead of source */ | |
457 static int Compare_Source_Map_by_channel(const void* a, const void* b) | |
458 { | |
459 return ( ((Source_Map*)a)->channel - ((Source_Map*)b)->channel ); | |
460 } | |
461 | |
462 /* Compare by albuffer */ | |
463 static int Compare_Buffer_Map(const void* a, const void* b) | |
464 { | |
1 | 465 return ( ((ALmixer_Buffer_Map*)a)->albuffer - ((ALmixer_Buffer_Map*)b)->albuffer ); |
0 | 466 } |
467 | |
468 /* This is for the user defined callback via | |
469 * ALmixer_ChannelFinished() | |
470 */ | |
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471 static void (*Channel_Done_Callback)(ALint which_channel, ALuint al_source, ALmixer_Data* almixer_data, ALboolean finished_naturally, void* user_data) = NULL; |
0 | 472 static void* Channel_Done_Callback_Userdata = NULL; |
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473 static void (*Channel_Data_Callback)(ALint which_channel, ALuint al_source, ALbyte* data, ALuint num_bytes, ALuint frequency, ALubyte channels, ALubyte bit_depth, ALboolean is_unsigned, ALboolean decode_mode_is_predecoded, ALuint length_in_msec, void* user_data) = NULL; |
1 | 474 static void* Channel_Data_Callback_Userdata = NULL; |
0 | 475 |
476 | |
477 static void PrintQueueStatus(ALuint source) | |
478 { | |
479 ALint buffers_queued = 0; | |
480 ALint buffers_processed = 0; | |
481 ALenum error; | |
482 | |
483 /* Get the number of buffers still queued */ | |
484 alGetSourcei( | |
485 source, | |
486 AL_BUFFERS_QUEUED, | |
487 &buffers_queued | |
488 ); | |
489 | |
490 if((error = alGetError()) != AL_NO_ERROR) | |
491 { | |
492 fprintf(stderr, "Error in PrintQueueStatus, Can't get buffers_queued: %s\n", | |
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493 alGetString(error)); |
0 | 494 } |
495 /* Get the number of buffers processed | |
496 * so we know if we need to refill | |
497 */ | |
498 alGetSourcei( | |
499 source, | |
500 AL_BUFFERS_PROCESSED, | |
501 &buffers_processed | |
502 ); | |
503 if((error = alGetError()) != AL_NO_ERROR) | |
504 { | |
505 fprintf(stderr, "Error in PrintQueueStatus, Can't get buffers_processed: %s\n", | |
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506 alGetString(error)); |
0 | 507 } |
508 | |
509 fprintf(stderr, "For source: %d, buffers_queued=%d, buffers_processed=%d\n", | |
510 source, | |
511 buffers_queued, | |
512 buffers_processed); | |
1 | 513 |
0 | 514 } |
515 | |
516 | |
517 | |
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518 static void Init_Channel(ALint channel) |
0 | 519 { |
520 | |
521 fprintf(stderr, "Init channel %d\n", channel); | |
522 ALmixer_Channel_List[channel].channel_in_use = 0; | |
523 ALmixer_Channel_List[channel].callback_update = 0; | |
524 ALmixer_Channel_List[channel].needs_stream = 0; | |
525 ALmixer_Channel_List[channel].paused = 0; | |
526 ALmixer_Channel_List[channel].halted = 0; | |
527 ALmixer_Channel_List[channel].loops = 0; | |
528 | |
529 ALmixer_Channel_List[channel].expire_ticks = 0; | |
530 ALmixer_Channel_List[channel].start_time = 0; | |
531 | |
532 ALmixer_Channel_List[channel].fade_enabled = 0; | |
533 ALmixer_Channel_List[channel].fade_expire_ticks = 0; | |
534 ALmixer_Channel_List[channel].fade_start_time = 0; | |
535 ALmixer_Channel_List[channel].fade_inv_time = 0.0f; | |
536 ALmixer_Channel_List[channel].fade_start_volume = 0.0f; | |
537 ALmixer_Channel_List[channel].fade_end_volume = 0.0f; | |
538 ALmixer_Channel_List[channel].max_volume = 1.0f; | |
539 ALmixer_Channel_List[channel].min_volume = 0.0f; | |
540 | |
541 ALmixer_Channel_List[channel].almixer_data = NULL; | |
542 } | |
543 /* Quick helper function to clean up a channel | |
544 * after it's done playing */ | |
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545 static void Clean_Channel(ALint channel) |
0 | 546 { |
547 ALenum error; | |
548 ALmixer_Channel_List[channel].channel_in_use = 0; | |
549 ALmixer_Channel_List[channel].callback_update = 0; | |
550 ALmixer_Channel_List[channel].needs_stream = 0; | |
551 ALmixer_Channel_List[channel].paused = 0; | |
552 ALmixer_Channel_List[channel].halted = 0; | |
553 ALmixer_Channel_List[channel].loops = 0; | |
554 | |
555 | |
556 ALmixer_Channel_List[channel].expire_ticks = 0; | |
557 ALmixer_Channel_List[channel].start_time = 0; | |
558 | |
559 ALmixer_Channel_List[channel].fade_enabled = 0; | |
560 ALmixer_Channel_List[channel].fade_expire_ticks = 0; | |
561 ALmixer_Channel_List[channel].fade_start_time = 0; | |
562 ALmixer_Channel_List[channel].fade_inv_time = 0.0f; | |
563 ALmixer_Channel_List[channel].fade_start_volume = 0.0f; | |
564 ALmixer_Channel_List[channel].fade_end_volume = 0.0f; | |
565 | |
566 alSourcef(ALmixer_Channel_List[channel].alsource, AL_MAX_GAIN, | |
567 ALmixer_Channel_List[channel].max_volume); | |
568 | |
569 if((error = alGetError()) != AL_NO_ERROR) | |
570 { | |
571 fprintf(stderr, "10Testing errpr before unqueue because getting stuff, for OS X this is expected: %s\n", | |
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572 alGetString(error)); |
0 | 573 } |
574 | |
575 alSourcef(ALmixer_Channel_List[channel].alsource, AL_MIN_GAIN, | |
576 ALmixer_Channel_List[channel].min_volume); | |
577 if((error = alGetError()) != AL_NO_ERROR) | |
578 { | |
579 fprintf(stderr, "11Testing errpr before unqueue because getting stuff, for OS X this is expected: %s\n", | |
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580 alGetString(error)); |
0 | 581 } |
582 | |
583 if(ALmixer_Channel_List[channel].almixer_data != NULL) | |
584 { | |
585 if(ALmixer_Channel_List[channel].almixer_data->in_use > 0) | |
586 { | |
587 ALmixer_Channel_List[channel].almixer_data->in_use--; | |
588 } | |
589 } | |
590 /* Needed to determine if rewind is needed, can't reset */ | |
591 /* | |
592 ALmixer_Channel_List[channel].almixer_data->eof = 0; | |
593 */ | |
594 | |
595 ALmixer_Channel_List[channel].almixer_data = NULL; | |
596 } | |
597 | |
598 | |
599 #if 0 | |
600 /* Not needed anymore because not doing any fileext checks. | |
601 * | |
602 * Unfortunately, strcasecmp isn't portable so here's a | |
603 * reimplementation of it (taken from SDL_sound) | |
604 */ | |
605 static int ALmixer_strcasecmp(const char* x, const char* y) | |
606 { | |
607 int ux, uy; | |
608 | |
609 if (x == y) /* same pointer? Both NULL? */ | |
610 return(0); | |
611 | |
612 if (x == NULL) | |
613 return(-1); | |
614 | |
615 if (y == NULL) | |
616 return(1); | |
617 | |
618 do | |
619 { | |
620 ux = toupper((int) *x); | |
621 uy = toupper((int) *y); | |
622 if (ux > uy) | |
623 return(1); | |
624 else if (ux < uy) | |
625 return(-1); | |
626 x++; | |
627 y++; | |
628 } while ((ux) && (uy)); | |
629 | |
630 return(0); | |
631 } | |
632 #endif | |
633 | |
634 | |
635 /* What shoud this return? | |
636 * 127 for signed, 255 for unsigned | |
637 */ | |
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638 static ALubyte GetSignednessValue(ALushort format) |
0 | 639 { |
640 switch(format) | |
641 { | |
642 case AUDIO_U8: | |
643 case AUDIO_U16LSB: | |
644 case AUDIO_U16MSB: | |
645 return ALMIXER_UNSIGNED_VALUE; | |
646 break; | |
647 case AUDIO_S8: | |
648 case AUDIO_S16LSB: | |
649 case AUDIO_S16MSB: | |
650 return ALMIXER_SIGNED_VALUE; | |
651 break; | |
652 default: | |
653 return 0; | |
654 } | |
655 return 0; | |
656 } | |
657 | |
658 | |
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659 static ALubyte GetBitDepth(ALushort format) |
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660 { |
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661 ALubyte bit_depth = 16; |
0 | 662 |
663 switch(format) | |
664 { | |
665 case AUDIO_U8: | |
666 case AUDIO_S8: | |
667 bit_depth = 8; | |
668 break; | |
669 | |
670 case AUDIO_U16LSB: | |
671 /* | |
672 case AUDIO_U16: | |
673 */ | |
674 case AUDIO_S16LSB: | |
675 /* | |
676 case AUDIO_S16: | |
677 */ | |
678 case AUDIO_U16MSB: | |
679 case AUDIO_S16MSB: | |
680 /* | |
681 case AUDIO_U16SYS: | |
682 case AUDIO_S16SYS: | |
683 */ | |
684 bit_depth = 16; | |
685 break; | |
686 default: | |
687 bit_depth = 0; | |
688 } | |
689 return bit_depth; | |
690 } | |
691 | |
692 /* Need to translate between SDL/SDL_Sound audiospec | |
693 * and OpenAL conventions */ | |
694 static ALenum TranslateFormat(Sound_AudioInfo* info) | |
695 { | |
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696 ALubyte bit_depth; |
0 | 697 |
698 bit_depth = GetBitDepth(info->format); | |
699 if(0 == bit_depth) | |
700 { | |
701 fprintf(stderr, "Warning: Unknown bit depth. Setting to 16\n"); | |
702 bit_depth = 16; | |
703 } | |
704 | |
705 if(2 == info->channels) | |
706 { | |
707 if(16 == bit_depth) | |
708 { | |
709 return AL_FORMAT_STEREO16; | |
710 } | |
711 else | |
712 { | |
713 return AL_FORMAT_STEREO8; | |
714 } | |
715 } | |
716 else | |
717 { | |
718 if(16 == bit_depth) | |
719 { | |
720 return AL_FORMAT_MONO16; | |
721 } | |
722 else | |
723 { | |
724 return AL_FORMAT_MONO8; | |
725 } | |
726 } | |
727 /* Make compiler happy. Shouldn't get here */ | |
728 return AL_FORMAT_STEREO16; | |
729 } | |
730 | |
1 | 731 |
732 /* This will compute the total playing time | |
733 * based upon the number of bytes and audio info. | |
734 * (In prinicple, it should compute the time for any given length) | |
735 */ | |
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736 static ALuint Compute_Total_Time_Decomposed(ALuint bytes_per_sample, ALuint frequency, ALubyte channels, size_t total_bytes) |
1 | 737 { |
738 double total_sec; | |
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739 ALuint total_msec; |
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740 ALuint bytes_per_sec; |
1 | 741 |
742 if(0 == total_bytes) | |
743 { | |
744 return 0; | |
745 } | |
746 /* To compute Bytes per second, do | |
747 * samples_per_sec * bytes_per_sample * number_of_channels | |
748 */ | |
749 bytes_per_sec = frequency * bytes_per_sample * channels; | |
750 | |
751 /* Now to get total time (sec), do | |
752 * total_bytes / bytes_per_sec | |
753 */ | |
754 total_sec = total_bytes / (double)bytes_per_sec; | |
755 | |
756 /* Now convert seconds to milliseconds | |
757 * Add .5 to the float to do rounding before the final cast | |
758 */ | |
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759 total_msec = (ALuint) ( (total_sec * 1000) + 0.5 ); |
1 | 760 /* |
761 fprintf(stderr, "freq=%d, bytes_per_sample=%d, channels=%d, total_msec=%d\n", frequency, bytes_per_sample, channels, total_msec); | |
762 */ | |
763 return total_msec; | |
764 } | |
765 | |
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766 static ALuint Compute_Total_Time(Sound_AudioInfo *info, size_t total_bytes) |
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767 { |
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768 ALuint bytes_per_sample; |
1 | 769 |
770 if(0 == total_bytes) | |
771 { | |
772 return 0; | |
773 } | |
774 /* SDL has a mask trick I was not aware of. Mask the upper bits | |
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775 * of the format, and you get 8 or 16 which is the bits per sample. |
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776 * Divide by 8bits_per_bytes and you get bytes_per_sample |
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777 * I tested this under 32-bit and 64-bit and big and little endian |
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778 * to make sure this still works since I have since moved from |
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779 * Uint32 to unspecified size types like ALuint. |
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780 */ |
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781 bytes_per_sample = (ALuint) ((info->format & 0xFF) / 8); |
1 | 782 |
783 return Compute_Total_Time_Decomposed(bytes_per_sample, info->rate, info->channels, total_bytes); | |
784 } /* End Compute_Total_Time */ | |
785 | |
786 | |
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787 static size_t Compute_Total_Bytes_Decomposed(ALuint bytes_per_sample, ALuint frequency, ALubyte channels, ALuint total_msec) |
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788 { |
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789 double total_sec; |
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790 ALuint bytes_per_sec; |
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791 size_t total_bytes; |
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792 |
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793 if(0 >= total_msec) |
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794 { |
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795 return 0; |
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796 } |
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797 /* To compute Bytes per second, do |
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798 * samples_per_sec * bytes_per_sample * number_of_channels |
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799 */ |
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800 bytes_per_sec = frequency * bytes_per_sample * channels; |
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801 |
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802 /* convert milliseconds to seconds */ |
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803 total_sec = total_msec / 1000.0; |
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804 |
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805 /* Now to get total bytes */ |
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806 total_bytes = (size_t)(((double)bytes_per_sec * total_sec) + 0.5); |
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807 |
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808 /* 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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809 */ |
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810 |
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811 return total_bytes; |
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812 } |
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813 |
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814 static size_t Compute_Total_Bytes(Sound_AudioInfo *info, ALuint total_msec) |
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815 { |
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816 ALuint bytes_per_sample; |
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817 |
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818 if(0 >= total_msec) |
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819 { |
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820 return 0; |
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821 } |
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822 /* SDL has a mask trick I was not aware of. Mask the upper bits |
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823 * of the format, and you get 8 or 16 which is the bits per sample. |
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824 * Divide by 8bits_per_bytes and you get bytes_per_sample |
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825 * I tested this under 32-bit and 64-bit and big and little endian |
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826 * to make sure this still works since I have since moved from |
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827 * Uint32 to unspecified size types like ALuint. |
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828 */ |
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829 bytes_per_sample = (ALuint) ((info->format & 0xFF) / 8); |
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830 |
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831 return Compute_Total_Bytes_Decomposed(bytes_per_sample, info->rate, info->channels, total_msec); |
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832 } |
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833 |
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834 /* The back-end decoders seem to need to decode in quantized frame sizes. |
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835 * So if I can pad the bytes to the next quanta, things might go more smoothly. |
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836 */ |
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837 static size_t Compute_Total_Bytes_With_Frame_Padding(Sound_AudioInfo *info, ALuint total_msec) |
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838 { |
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839 ALuint bytes_per_sample; |
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840 ALuint bytes_per_frame; |
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841 size_t evenly_divisible_frames; |
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842 size_t remainder_frames; |
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843 size_t return_bytes; |
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844 |
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845 size_t total_bytes = Compute_Total_Bytes(info, total_msec); |
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846 |
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847 bytes_per_sample = (ALuint) ((info->format & 0xFF) / 8); |
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848 |
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849 bytes_per_frame = bytes_per_sample * info->channels; |
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850 |
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851 evenly_divisible_frames = total_bytes / bytes_per_frame; |
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852 remainder_frames = total_bytes % bytes_per_frame; |
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853 |
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854 return_bytes = (evenly_divisible_frames * bytes_per_frame) + (remainder_frames * bytes_per_frame); |
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855 |
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856 /* Experimentally, some times I see to come up short in |
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857 * actual bytes decoded and I see a second pass is needed. |
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858 * I'm worried this may have additional performance implications. |
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859 * Sometimes in the second pass (depending on file), |
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860 * I have seen between 0 and 18 bytes. |
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861 * I'm tempted to pad the bytes by some arbitrary amount. |
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862 * However, I think currently the way SDL_sound is implemented, |
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863 * there is a big waste of memory up front instead of per-pass, |
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864 * so maybe I shouldn't worry about this. |
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865 */ |
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866 /* |
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867 return_bytes += 64; |
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868 */ |
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869 /* |
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870 fprintf(stderr, "remainder_frames=%d, padded_total_bytes=%d\n", remainder_frames, return_bytes); |
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871 */ |
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872 return return_bytes; |
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873 |
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874 } |
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875 |
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876 |
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877 |
1 | 878 |
0 | 879 /**************** REMOVED ****************************/ |
880 /* This was removed because I originally thought | |
881 * OpenAL could return a pointer to the buffer data, | |
882 * but I was wrong. If something like that is ever | |
883 * implemented, then this might become useful. | |
884 */ | |
885 #if 0 | |
886 /* Reconstruct_Sound_Sample and Set_AudioInfo only | |
887 * are needed if the Seek memory optimization is | |
888 * used. Also, the Loki dist doesn't seem to support | |
889 * AL_DATA which I need for it. | |
890 */ | |
891 #ifndef DISABLE_SEEK_MEMORY_OPTIMIZATION | |
892 | |
893 static void Set_AudioInfo(Sound_AudioInfo* info, ALint frequency, ALint bits, ALint channels) | |
894 { | |
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895 info->rate = (ALuint)frequency; |
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896 info->channels = (ALubyte)channels; |
0 | 897 |
898 /* Not sure if it should be signed or unsigned. Hopefully | |
899 * that detail won't be needed. | |
900 */ | |
901 if(8 == bits) | |
902 { | |
903 info->format = AUDIO_U8; | |
904 } | |
905 else | |
906 { | |
907 info->format = AUDIO_U16SYS; | |
908 } | |
909 fprintf(stderr, "Audio info: freq=%d, chan=%d, format=%d\n", | |
910 info->rate, info->channels, info->format); | |
911 | |
912 } | |
913 | |
914 | |
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915 static ALint Reconstruct_Sound_Sample(ALmixer_Data* data) |
0 | 916 { |
917 ALenum error; | |
918 ALint* data_from_albuffer; | |
919 ALint freq; | |
920 ALint bits; | |
921 ALint channels; | |
922 ALint size; | |
923 | |
924 /* Create memory all initiallized to 0. */ | |
925 data->sample = (Sound_Sample*)calloc(1, sizeof(Sound_Sample)); | |
926 if(NULL == data->sample) | |
927 { | |
928 ALmixer_SetError("Out of memory for Sound_Sample"); | |
929 return -1; | |
930 } | |
931 | |
932 /* Clear errors */ | |
933 alGetError(); | |
934 | |
935 alGetBufferi(data->buffer[0], AL_FREQUENCY, &freq); | |
936 if((error = alGetError()) != AL_NO_ERROR) | |
937 { | |
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938 ALmixer_SetError("alGetBufferi(AL_FREQUENCY): %s", alGetString(error) ); |
0 | 939 free(data->sample); |
940 data->sample = NULL; | |
941 return -1; | |
942 } | |
943 | |
944 alGetBufferi(data->buffer[0], AL_BITS, &bits); | |
945 if((error = alGetError()) != AL_NO_ERROR) | |
946 { | |
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947 ALmixer_SetError("alGetBufferi(AL_BITS): %s", alGetString(error) ); |
0 | 948 free(data->sample); |
949 data->sample = NULL; | |
950 return -1; | |
951 } | |
952 | |
953 alGetBufferi(data->buffer[0], AL_CHANNELS, &channels); | |
954 if((error = alGetError()) != AL_NO_ERROR) | |
955 { | |
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956 ALmixer_SetError("alGetBufferi(AL_CHANNELS): %s", alGetString(error) ); |
0 | 957 free(data->sample); |
958 data->sample = NULL; | |
959 return -1; | |
960 } | |
961 | |
962 alGetBufferi(data->buffer[0], AL_SIZE, &size); | |
963 if((error = alGetError()) != AL_NO_ERROR) | |
964 { | |
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965 ALmixer_SetError("alGetBufferi(AL_SIZE): %s", alGetString(error) ); |
0 | 966 free(data->sample); |
967 data->sample = NULL; | |
968 return -1; | |
969 } | |
970 | |
971 alGetBufferi(data->buffer[0], AL_DATA, data_from_albuffer); | |
972 if((error = alGetError()) != AL_NO_ERROR) | |
973 { | |
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974 ALmixer_SetError("alGetBufferi(AL_DATA): %s", alGetString(error) ); |
0 | 975 free(data->sample); |
976 data->sample = NULL; | |
977 return -1; | |
978 } | |
979 | |
980 if(size <= 0) | |
981 { | |
982 ALmixer_SetError("No data in al buffer"); | |
983 free(data->sample); | |
984 data->sample = NULL; | |
985 return -1; | |
986 } | |
987 | |
988 /* Now that we have all the attributes, we need to | |
989 * allocate memory for the buffer and reconstruct | |
990 * the AudioInfo attributes. | |
991 */ | |
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992 data->sample->buffer = malloc(size*sizeof(ALbyte)); |
0 | 993 if(NULL == data->sample->buffer) |
994 { | |
995 ALmixer_SetError("Out of memory for sample->buffer"); | |
996 free(data->sample); | |
997 data->sample = NULL; | |
998 return -1; | |
999 } | |
1000 | |
1001 memcpy(data->sample->buffer, data_from_albuffer, size); | |
1002 data->sample->buffer_size = size; | |
1003 | |
1004 /* Fill up the Sound_AudioInfo structures */ | |
1005 Set_AudioInfo(&data->sample->desired, freq, bits, channels); | |
1006 Set_AudioInfo(&data->sample->actual, freq, bits, channels); | |
1007 | |
1008 return 0; | |
1009 } | |
1010 | |
1011 #endif /* End DISABLE_SEEK_MEMORY_OPTIMIZATION */ | |
1012 #endif | |
1013 /*************** END REMOVED *************************/ | |
1014 | |
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1015 static void Invoke_Channel_Done_Callback(ALint which_channel, ALboolean did_finish_naturally) |
0 | 1016 { |
1017 if(NULL == Channel_Done_Callback) | |
1018 { | |
1019 return; | |
1020 } | |
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1021 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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1022 } |
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1023 |
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1024 static ALint LookUpBuffer(ALuint buffer, ALmixer_Buffer_Map* buffer_map_list, ALuint num_items_in_list) |
0 | 1025 { |
1026 /* Only the first value is used for the key */ | |
1 | 1027 ALmixer_Buffer_Map key = { 0, 0, NULL, 0 }; |
1028 ALmixer_Buffer_Map* found_item = NULL; | |
0 | 1029 key.albuffer = buffer; |
1030 | |
1031 /* Use the ANSI C binary search feature (yea!) */ | |
1 | 1032 found_item = (ALmixer_Buffer_Map*)bsearch(&key, buffer_map_list, num_items_in_list, sizeof(ALmixer_Buffer_Map), Compare_Buffer_Map); |
0 | 1033 if(NULL == found_item) |
1034 { | |
1035 ALmixer_SetError("Can't find buffer"); | |
1036 return -1; | |
1037 } | |
1038 return found_item->index; | |
1039 } | |
1040 | |
1041 | |
1042 /* FIXME: Need to pass back additional info to be useful. | |
1043 * Bit rate, stereo/mono (num chans), time in msec? | |
1044 * Precoded/streamed flag so user can plan for future data? | |
1045 */ | |
1 | 1046 /* |
1047 * channels: 1 for mono, 2 for stereo | |
1048 * | |
1049 */ | |
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1050 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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1051 { |
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1052 ALboolean is_unsigned; |
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1053 ALubyte bits_per_sample = GetBitDepth(format); |
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1054 ALuint bytes_per_sample; |
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1055 ALuint length_in_msec; |
1 | 1056 |
1057 if(GetSignednessValue(format) == ALMIXER_UNSIGNED_VALUE) | |
1058 { | |
1059 is_unsigned = 1; | |
1060 } | |
1061 else | |
1062 { | |
1063 is_unsigned = 0; | |
1064 } | |
1065 | |
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1066 bytes_per_sample = (ALuint) (bits_per_sample / 8); |
1 | 1067 |
1068 length_in_msec = Compute_Total_Time_Decomposed(bytes_per_sample, frequency, channels, num_bytes); | |
1069 | |
0 | 1070 /* |
1071 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); | |
1072 */ | |
1073 if(NULL == Channel_Data_Callback) | |
1074 { | |
1075 return; | |
1076 } | |
1 | 1077 /* |
1078 * Channel_Data_Callback(which_channel, data, num_bytes, frequency, channels, GetBitDepth(format), format, decode_mode_is_predecoded); | |
1079 */ | |
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1080 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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1081 } |
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1082 |
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1083 static void Invoke_Predecoded_Channel_Data_Callback(ALint channel, ALmixer_Data* data) |
0 | 1084 { |
1085 if(NULL == data->sample) | |
1086 { | |
1087 return; | |
1088 } | |
1089 /* The buffer position is complicated because if the current data was seeked, | |
1090 * we must adjust the buffer to the seek position | |
1091 */ | |
1092 Invoke_Channel_Data_Callback(channel, | |
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1093 (((ALbyte*) data->sample->buffer) + (data->total_bytes - data->loaded_bytes) ), |
0 | 1094 data->loaded_bytes, |
1095 data->sample->desired.rate, | |
1096 data->sample->desired.channels, | |
1097 data->sample->desired.format, | |
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1098 AL_TRUE |
0 | 1099 ); |
1100 } | |
1101 | |
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1102 static void Invoke_Streamed_Channel_Data_Callback(ALint channel, ALmixer_Data* data, ALuint buffer) |
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1103 { |
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1104 ALint index; |
0 | 1105 if(NULL == data->buffer_map_list) |
1106 { | |
1107 return; | |
1108 } | |
1109 index = LookUpBuffer(buffer, data->buffer_map_list, data->max_queue_buffers); | |
1110 /* This should catch the case where all buffers are unqueued | |
1111 * and the "current" buffer is id: 0 | |
1112 */ | |
1113 if(-1 == index) | |
1114 { | |
1115 return; | |
1116 } | |
1117 Invoke_Channel_Data_Callback(channel, | |
1118 data->buffer_map_list[index].data, | |
1119 data->buffer_map_list[index].num_bytes, | |
1120 data->sample->desired.rate, | |
1121 data->sample->desired.channels, | |
1122 data->sample->desired.format, | |
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1123 AL_FALSE |
0 | 1124 ); |
1125 } | |
1126 | |
1127 /* From SDL_Sound's playsound. Converts milliseconds to byte positions. | |
1128 * This is needed for seeking on predecoded samples | |
1129 */ | |
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1130 static ALuint Convert_Msec_To_Byte_Pos(Sound_AudioInfo *info, ALuint ms) |
0 | 1131 { |
1132 float frames_per_ms; | |
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1133 ALuint frame_offset; |
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1134 ALuint frame_size; |
0 | 1135 fprintf(stderr, "In convert\n" ); |
1136 if(info == NULL) | |
1137 { | |
1138 fprintf(stderr, "Error, info is NULL\n"); | |
1139 } | |
1140 else | |
1141 { | |
1142 fprintf(stderr, "Not an error: info is not NULL\n"); | |
1143 } | |
1144 fprintf(stderr, "The rate=%d\n", info->rate); | |
1145 | |
1146 /* "frames" == "sample frames" */ | |
1147 frames_per_ms = ((float) info->rate) / 1000.0f; | |
1148 fprintf(stderr, "%f\n", frames_per_ms); | |
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1149 frame_offset = (ALuint) (frames_per_ms * ((float) ms)); |
0 | 1150 fprintf(stderr, "%d\n", frame_offset); |
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1151 frame_size = (ALuint) ((info->format & 0xFF) / 8) * info->channels; |
0 | 1152 fprintf(stderr, "%d\n", frame_size); |
1153 return(frame_offset * frame_size); | |
1154 } /* cvtMsToBytePos */ | |
1155 | |
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1156 static ALint Set_Predecoded_Seek_Position(ALmixer_Data* data, ALuint byte_position) |
0 | 1157 { |
1158 ALenum error; | |
1159 /* clear error */ | |
1160 alGetError(); | |
1161 | |
1162 /* Is it greater than, or greater-than or equal to ?? */ | |
1163 if(byte_position > data->total_bytes) | |
1164 { | |
1165 /* We can't go past the end, so set to end? */ | |
1166 fprintf(stderr, "Error, can't seek past end\n"); | |
1167 | |
1168 /* In case the below thing doesn't work, | |
1169 * just rewind the whole thing. | |
1170 * | |
1171 alBufferData(data->buffer[0], | |
1172 TranslateFormat(&data->sample->desired), | |
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1173 (ALbyte*) data->sample->buffer, |
0 | 1174 data->total_bytes, |
1175 data->sample->desired.rate | |
1176 ); | |
1177 */ | |
1178 | |
1179 /* I was trying to set to the end, (1 byte remaining), | |
1180 * but I was getting freezes. I'm thinking it might be | |
1181 * another Power of 2 bug in the Loki dist. I tried 2, | |
1182 * and it still hung. 4 didn't hang, but I got a clip | |
1183 * artifact. 8 seemed to work okay. | |
1184 */ | |
1185 alBufferData(data->buffer[0], | |
1186 TranslateFormat(&data->sample->desired), | |
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1187 (((ALbyte*) data->sample->buffer) + (data->total_bytes - 8) ), |
0 | 1188 8, |
1189 data->sample->desired.rate | |
1190 ); | |
1191 if( (error = alGetError()) != AL_NO_ERROR) | |
1192 { | |
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1193 ALmixer_SetError("Can't seek past end and alBufferData failed: %s\n", alGetString(error)); |
0 | 1194 return -1; |
1195 } | |
1196 /* Need to set the loaded_bytes field because I don't trust the OpenAL | |
1197 * query command to work because I don't know if it will mutilate the | |
1198 * size for its own purposes or return the original size | |
1199 */ | |
1200 data->loaded_bytes = 8; | |
1201 | |
1202 /* Not sure if this should be an error or not */ | |
1203 /* | |
1204 ALmixer_SetError("Can't Seek past end"); | |
1205 return -1; | |
1206 */ | |
1207 return 0; | |
1208 } | |
1209 | |
1210 alBufferData(data->buffer[0], | |
1211 TranslateFormat(&data->sample->desired), | |
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1212 &(((ALbyte*)data->sample->buffer)[byte_position]), |
0 | 1213 data->total_bytes - byte_position, |
1214 data->sample->desired.rate | |
1215 ); | |
1216 if( (error = alGetError()) != AL_NO_ERROR) | |
1217 { | |
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1218 ALmixer_SetError("alBufferData failed: %s\n", alGetString(error)); |
0 | 1219 return -1; |
1220 } | |
1221 /* Need to set the loaded_bytes field because I don't trust the OpenAL | |
1222 * query command to work because I don't know if it will mutilate the | |
1223 * size for its own purposes or return the original size | |
1224 */ | |
1225 data->loaded_bytes = data->total_bytes - byte_position; | |
1226 | |
1227 return 0; | |
1228 } | |
1229 | |
1230 /* Because we have multiple queue buffers and OpenAL won't let | |
1231 * us access them, we need to keep copies of each buffer around | |
1232 */ | |
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1233 static ALint CopyDataToAccessBuffer(ALmixer_Data* data, ALuint num_bytes, ALuint buffer) |
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1234 { |
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1235 ALint index; |
0 | 1236 /* We only want to copy if access_data is true. |
1237 * This is determined by whether memory has been | |
1238 * allocated in the buffer_map_list or not | |
1239 */ | |
1240 if(NULL == data->buffer_map_list) | |
1241 { | |
1242 return -1; | |
1243 } | |
1244 index = LookUpBuffer(buffer, data->buffer_map_list, data->max_queue_buffers); | |
1245 if(-1 == index) | |
1246 { | |
1247 fprintf(stderr, ">>>>>>>CopyData catch, albuffer=%d\n",buffer); | |
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 */ | |
1385 fprintf(stderr, "ALTERED bytes decoded for silence: Original end was %d\n", bytes_decoded); | |
1386 bytes_decoded = data->sample->buffer_size; | |
1387 } | |
1388 /*********** END EXPERIMENT ******************************/ | |
1389 /******* END REMOVE ME ********************************/ | |
1390 #endif | |
1391 | |
1392 /* Now copy the data to the OpenAL buffer */ | |
1393 /* We can't just set a pointer because the API needs | |
1394 * its own copy to assist hardware acceleration */ | |
1395 alBufferData(buffer, | |
1396 TranslateFormat(&data->sample->desired), | |
1397 data->sample->buffer, | |
1398 bytes_decoded, | |
1399 data->sample->desired.rate | |
1400 ); | |
1401 if( (error = alGetError()) != AL_NO_ERROR) | |
1402 { | |
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1403 ALmixer_SetError("alBufferData failed: %s\n", alGetString(error)); |
0 | 1404 return 0; |
1405 } | |
1406 | |
1407 /* If we need to, copy the data also to the access area | |
1408 * (the function will do the check for us) | |
1409 */ | |
1410 CopyDataToAccessBuffer(data, bytes_decoded, buffer); | |
1411 return bytes_decoded; | |
1412 } | |
1413 | |
1414 | |
1415 | |
1416 | |
1417 /******************** EXPERIEMENT **************************** | |
1418 * Test function to force maximum buffer filling during loops | |
1419 * REMOVE LATER | |
1420 *********************************************/ | |
1421 #if 0 | |
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1422 static ALint GetMoreData2(ALmixer_Data* data, ALuint buffer) |
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1423 { |
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1424 ALint bytes_decoded; |
0 | 1425 ALenum error; |
1426 if(NULL == data) | |
1427 { | |
1428 ALmixer_SetError("Cannot GetMoreData() because ALmixer_Data* is NULL\n"); | |
1429 return -1; | |
1430 } | |
1431 | |
1432 if(AL_FALSE == alIsBuffer(buffer)) | |
1433 { | |
1434 fprintf(stderr, "NOT A BUFFER>>>>>>>>>>>>>>>\n"); | |
1435 return -1; | |
1436 } | |
1437 fprintf(stderr, "Entered GetMoreData222222: buffer id is %d\n", buffer); | |
1438 | |
1439 /* | |
1440 fprintf(stderr, "Decode in GetMoreData\n"); | |
1441 */ | |
1442 | |
1443 #if 0 | |
1444 if(buffer%2 == 1) | |
1445 { | |
1446 fprintf(stderr, "Setting buffer size to 16384\n"); | |
1447 Sound_SetBufferSize(data->sample, 16384); | |
1448 } | |
1449 else | |
1450 { | |
1451 fprintf(stderr, "Setting buffer size to 8192\n"); | |
1452 Sound_SetBufferSize(data->sample, 8192); | |
1453 } | |
1454 #endif | |
1455 | |
1456 bytes_decoded = Sound_Decode(data->sample); | |
1457 if(data->sample->flags & SOUND_SAMPLEFLAG_ERROR) | |
1458 { | |
1459 fprintf(stderr, "Sound_Decode triggered an ERROR>>>>>>\n"); | |
1460 ALmixer_SetError(Sound_GetError()); | |
1461 /* | |
1462 Sound_FreeSample(data->sample); | |
1463 */ | |
1464 return -1; | |
1465 } | |
1466 /* Don't forget to add check for EOF */ | |
1467 /* Will return 0 bytes and pass the buck to check sample->flags */ | |
1468 if(0 == bytes_decoded) | |
1469 { | |
1470 #if 1 | |
1471 fprintf(stderr, "Hit eof while trying to buffer\n"); | |
1472 data->eof = 1; | |
1473 if(data->sample->flags & SOUND_SAMPLEFLAG_EOF) | |
1474 { | |
1475 fprintf(stderr, "\tEOF flag\n"); | |
1476 } | |
1477 if(data->sample->flags & SOUND_SAMPLEFLAG_CANSEEK) | |
1478 { | |
1479 fprintf(stderr, "\tCanSeek flag\n"); | |
1480 } | |
1481 if(data->sample->flags & SOUND_SAMPLEFLAG_EAGAIN) | |
1482 { | |
1483 fprintf(stderr, "\tEAGAIN flag\n"); | |
1484 } | |
1485 if(data->sample->flags & SOUND_SAMPLEFLAG_NONE) | |
1486 { | |
1487 fprintf(stderr, "\tNONE flag\n"); | |
1488 } | |
1489 #endif | |
1490 return 0; | |
1491 } | |
1492 | |
1493 if(bytes_decoded < 16384) | |
1494 { | |
1495 char* tempbuffer1 = (char*)malloc(16384); | |
1496 char* tempbuffer2 = (char*)malloc(16384); | |
1497 int retval; | |
1498 memcpy(tempbuffer1, data->sample->buffer, bytes_decoded); | |
1499 retval = Sound_SetBufferSize(data->sample, 16384-bytes_decoded); | |
1500 if(retval == 1) | |
1501 { | |
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1502 ALuint new_bytes; |
0 | 1503 Sound_Rewind(data->sample); |
1504 new_bytes = Sound_Decode(data->sample); | |
1505 fprintf(stderr, "Orig bytes: %d, Make up bytes_decoded=%d, total=%d\n", bytes_decoded, new_bytes, new_bytes+bytes_decoded); | |
1506 | |
1507 memcpy(tempbuffer2, data->sample->buffer, new_bytes); | |
1508 | |
1509 retval = Sound_SetBufferSize(data->sample, 16384); | |
1510 fprintf(stderr, "Finished reset...now danger copy\n"); | |
1511 memcpy(data->sample->buffer, tempbuffer1,bytes_decoded); | |
1512 | |
1513 fprintf(stderr, "Finished reset...now danger copy2\n"); | |
1514 memcpy( &( ((char*)(data->sample->buffer))[bytes_decoded] ), tempbuffer2, new_bytes); | |
1515 | |
1516 fprintf(stderr, "Finished \n"); | |
1517 | |
1518 free(tempbuffer1); | |
1519 free(tempbuffer2); | |
1520 bytes_decoded += new_bytes; | |
1521 fprintf(stderr, "ASSERT bytes should equal 16384: %d\n", bytes_decoded); | |
1522 } | |
1523 else | |
1524 { | |
1525 fprintf(stderr, "Experiment failed: %s\n", Sound_GetError()); | |
1526 } | |
1527 } | |
1528 | |
1529 /* Now copy the data to the OpenAL buffer */ | |
1530 /* We can't just set a pointer because the API needs | |
1531 * its own copy to assist hardware acceleration */ | |
1532 alBufferData(buffer, | |
1533 TranslateFormat(&data->sample->desired), | |
1534 data->sample->buffer, | |
1535 bytes_decoded, | |
1536 data->sample->desired.rate | |
1537 ); | |
1538 if( (error = alGetError()) != AL_NO_ERROR) | |
1539 { | |
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1540 ALmixer_SetError("alBufferData failed: %s\n", alGetString(error)); |
0 | 1541 return -1; |
1542 } | |
1543 | |
1544 fprintf(stderr, "GetMoreData2222 returning %d bytes decoded\n", bytes_decoded); | |
1545 return bytes_decoded; | |
1546 } | |
1547 #endif | |
1548 | |
1549 /************ END EXPERIEMENT - REMOVE ME *************************/ | |
1550 | |
1551 | |
1552 | |
1553 | |
1554 | |
1555 | |
1556 | |
1557 | |
1558 | |
1559 /* This function will look up the source for the corresponding channel */ | |
1560 /* Must return 0 on error instead of -1 because of unsigned int */ | |
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1561 static ALuint Internal_GetSource(ALint channel) |
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1562 { |
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1563 ALint i; |
0 | 1564 /* Make sure channel is in bounds */ |
1565 if(channel >= Number_of_Channels_global) | |
1566 { | |
1567 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); | |
1568 return 0; | |
1569 } | |
1570 /* If the user specified -1, then return the an available source */ | |
1571 if(channel < 0) | |
1572 { | |
1573 for(i=Number_of_Reserve_Channels_global; i<Number_of_Channels_global; i++) | |
1574 { | |
1575 if( ! ALmixer_Channel_List[i].channel_in_use ) | |
1576 { | |
1577 return ALmixer_Channel_List[i].alsource; | |
1578 } | |
1579 } | |
1580 /* If we get here, all sources are in use */ | |
1581 /* Error message seems too harsh | |
1582 ALmixer_SetError("All sources are in use"); | |
1583 */ | |
1584 return 0; | |
1585 } | |
1586 /* Last case: Return the source for the channel */ | |
1587 return ALmixer_Channel_List[channel].alsource; | |
1588 } | |
1589 | |
1590 /* This function will look up the channel for the corresponding source */ | |
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1591 static ALint Internal_GetChannel(ALuint source) |
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1592 { |
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1593 ALint i; |
0 | 1594 /* Only the first value is used for the key */ |
1595 Source_Map key = { 0, 0 }; | |
1596 Source_Map* found_item = NULL; | |
1597 key.source = source; | |
1598 | |
1599 /* If the source is 0, look up the first available channel */ | |
1600 if(0 == source) | |
1601 { | |
1602 for(i=Number_of_Reserve_Channels_global; i<Number_of_Channels_global; i++) | |
1603 { | |
1604 if( ! ALmixer_Channel_List[i].channel_in_use ) | |
1605 { | |
1606 return i; | |
1607 } | |
1608 } | |
1609 /* If we get here, all sources are in use */ | |
1610 /* Error message seems too harsh | |
1611 ALmixer_SetError("All channels are in use"); | |
1612 */ | |
1613 return -1; | |
1614 } | |
1615 | |
1616 | |
1617 /* Else, look up the source and return the channel */ | |
1618 if(AL_FALSE == alIsSource(source)) | |
1619 { | |
1620 ALmixer_SetError("Is not a source"); | |
1621 return -1; | |
1622 } | |
1623 | |
1624 /* Use the ANSI C binary search feature (yea!) */ | |
1625 found_item = (Source_Map*)bsearch(&key, Source_Map_List, Number_of_Channels_global, sizeof(Source_Map), Compare_Source_Map); | |
1626 if(NULL == found_item) | |
1627 { | |
1628 ALmixer_SetError("Source is valid but not registered with ALmixer (to a channel)"); | |
1629 return -1; | |
1630 } | |
1631 return found_item->channel; | |
1632 } | |
1633 | |
1634 | |
1635 | |
1636 /* This function will find the first available channel (not in use) | |
1637 * from the specified start channel. Reserved channels to not qualify | |
1638 * as available. | |
1639 */ | |
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1640 static ALint Internal_FindFreeChannel(ALint start_channel) |
0 | 1641 { |
1642 /* Start at the number of reserved so we skip over | |
1643 * all the reserved channels. | |
1644 */ | |
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1645 ALint i = Number_of_Reserve_Channels_global; |
0 | 1646 /* Quick check to see if we're out of bounds */ |
1647 if(start_channel >= Number_of_Channels_global) | |
1648 { | |
1649 return -1; | |
1650 } | |
1651 | |
1652 /* If the start channel is even higher than the reserved, | |
1653 * then start at the higher value. | |
1654 */ | |
1655 if(start_channel > Number_of_Reserve_Channels_global) | |
1656 { | |
1657 i = start_channel; | |
1658 } | |
1659 | |
1660 /* i has already been set */ | |
1661 for( ; i<Number_of_Channels_global; i++) | |
1662 { | |
1663 if( ! ALmixer_Channel_List[i].channel_in_use ) | |
1664 { | |
1665 return i; | |
1666 } | |
1667 } | |
1668 /* If we get here, all sources are in use */ | |
1669 return -1; | |
1670 } | |
1671 | |
1672 | |
1673 | |
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1674 /* Will return the number of channels halted |
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1675 * or 0 for error |
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1676 */ |
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1677 static ALint Internal_HaltChannel(ALint channel, ALboolean did_finish_naturally) |
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1678 { |
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1679 ALint retval = 0; |
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1680 ALint counter = 0; |
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1681 ALenum error; |
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1682 ALint buffers_still_queued; |
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1683 ALint buffers_processed; |
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1684 |
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1685 if(channel >= Number_of_Channels_global) |
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1686 { |
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1687 ALmixer_SetError("Cannot halt channel %d because it exceeds maximum number of channels (%d)\n", channel, Number_of_Channels_global); |
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1688 return -1; |
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1689 } |
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1690 /* If the user specified a specific channel */ |
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1691 if(channel >= 0) |
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1692 { |
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1693 fprintf(stderr, "Halt on channel %d\n", channel); |
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1694 /* only need to process channel if in use */ |
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1695 if(ALmixer_Channel_List[channel].channel_in_use) |
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1696 { |
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1697 alSourceStop(ALmixer_Channel_List[channel].alsource); |
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1698 if((error = alGetError()) != AL_NO_ERROR) |
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1699 { |
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1700 fprintf(stderr, "14Testing error: %s\n", |
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1701 alGetString(error)); |
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1702 } |
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1703 /* Here's the situation. My old method of using |
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1704 * alSourceUnqueueBuffers() seemed to be invalid in light |
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1705 * of all the problems I suffered through with getting |
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1706 * the CoreData backend to work with this code. |
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1707 * As such, I'm changing all the code to set the buffer to |
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1708 * AL_NONE. Furthermore, the queued vs. non-queued issue |
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1709 * doesn't need to apply here. For non-queued, Loki, |
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1710 * Creative Windows, and CoreAudio seem to leave the |
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1711 * buffer queued (Old Mac didn't.) For queued, we need to |
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1712 * remove the processed buffers and force remove the |
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1713 * still-queued buffers. |
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1714 */ |
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1715 fprintf(stderr, "Halt on channel %d, channel in use\n", channel); |
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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 Clean_Channel(channel); |
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1761 Is_Playing_global--; |
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1762 /* Launch callback for consistency? */ |
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1763 Invoke_Channel_Done_Callback(channel, did_finish_naturally); |
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1764 counter++; |
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1765 } |
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1766 } |
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1767 /* The user wants to halt all channels */ |
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1768 else |
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1769 { |
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1770 ALint i; |
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1771 for(i=0; i<Number_of_Channels_global; i++) |
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1772 { |
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1773 fprintf(stderr, "Halting channel %d\n", i); |
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1774 fprintf(stderr, "in use %d\n", ALmixer_Channel_List[i].channel_in_use ); |
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1775 /* only need to process channel if in use */ |
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1776 if(ALmixer_Channel_List[i].channel_in_use) |
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1777 { |
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1778 fprintf(stderr, "SourceStop %d\n", i); |
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1779 alSourceStop(ALmixer_Channel_List[i].alsource); |
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1780 if((error = alGetError()) != AL_NO_ERROR) |
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1781 { |
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1782 fprintf(stderr, "19Testing error: %s\n", |
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1783 alGetString(error)); |
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1784 } |
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1785 |
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1786 /* Here's the situation. My old method of using |
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1787 * alSourceUnqueueBuffers() seemed to be invalid in light |
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1788 * of all the problems I suffered through with getting |
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1789 * the CoreData backend to work with this code. |
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1790 * As such, I'm changing all the code to set the buffer to |
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1791 * AL_NONE. Furthermore, the queued vs. non-queued issue |
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1792 * doesn't need to apply here. For non-queued, Loki, |
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1793 * Creative Windows, and CoreAudio seem to leave the |
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1794 * buffer queued (Old Mac didn't.) For queued, we need to |
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1795 * remove the processed buffers and force remove the |
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1796 * still-queued buffers. |
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1797 */ |
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1798 fprintf(stderr, "Halt on channel %d, channel in use\n", channel); |
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1799 alGetSourcei( |
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1800 ALmixer_Channel_List[i].alsource, |
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1801 AL_BUFFERS_QUEUED, &buffers_still_queued |
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1802 ); |
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1803 if((error = alGetError()) != AL_NO_ERROR) |
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1804 { |
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1805 fprintf(stderr, "17Testing Error with buffers_still_queued: %s", |
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1806 alGetString(error)); |
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1807 ALmixer_SetError("Failed detecting still queued buffers: %s", |
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1808 alGetString(error) ); |
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1809 retval = -1; |
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1810 } |
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1811 alGetSourcei( |
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1812 ALmixer_Channel_List[i].alsource, |
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1813 AL_BUFFERS_PROCESSED, &buffers_processed |
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1814 ); |
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1815 if((error = alGetError()) != AL_NO_ERROR) |
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1816 { |
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1817 fprintf(stderr, "17Testing Error with buffers_processed: %s", |
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1818 alGetString(error)); |
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1819 ALmixer_SetError("Failed detecting still processed buffers: %s", |
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1820 alGetString(error) ); |
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1821 retval = -1; |
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1822 } |
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1823 /* If either of these is greater than 0, it means we need |
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1824 * to clear the source |
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1825 */ |
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1826 if((buffers_still_queued > 0) || (buffers_processed > 0)) |
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1827 { |
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1828 alSourcei(ALmixer_Channel_List[i].alsource, |
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1829 AL_BUFFER, |
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1830 AL_NONE); |
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1831 if((error = alGetError()) != AL_NO_ERROR) |
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1832 { |
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1833 fprintf(stderr, "17Testing Error with clearing buffer from source: %s", |
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1834 alGetString(error)); |
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1835 ALmixer_SetError("Failed to clear buffer from source: %s", |
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1836 alGetString(error) ); |
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1837 retval = -1; |
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1838 } |
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1839 } |
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1840 |
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1841 ALmixer_Channel_List[i].almixer_data->num_buffers_in_use = 0; |
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1842 |
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1843 fprintf(stderr, "Clean channel %d\n", i); |
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1844 Clean_Channel(i); |
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1845 Is_Playing_global--; |
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1846 /* Launch callback for consistency? */ |
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1847 fprintf(stderr, "Callback%d\n", i); |
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1848 Invoke_Channel_Done_Callback(i, did_finish_naturally); |
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1849 |
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1850 /* Increment the counter */ |
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1851 counter++; |
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1852 } |
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1853 /* Let's halt everything just in case there |
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1854 * are bugs. |
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1855 */ |
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1856 /* |
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1857 else |
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1858 { |
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1859 alSourceStop(ALmixer_Channel_List[channel].alsource); |
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1860 / * Can't clean because the in_use counter for |
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1861 * data will get messed up * / |
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1862 Clean_Channel(channel); |
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1863 } |
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1864 */ |
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1865 /* Just in case */ |
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1866 Is_Playing_global = 0; |
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1867 } |
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1868 } |
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1869 if(-1 == retval) |
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1870 { |
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1871 return -1; |
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1872 } |
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1873 return counter; |
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1874 } |
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1875 |
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1876 |
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1877 /* Will return the source halted or the total number of channels |
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1878 * if all were halted or 0 for error |
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1879 */ |
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1880 static ALint Internal_HaltSource(ALuint source, ALboolean did_finish_naturally) |
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1881 { |
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1882 ALint channel; |
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1883 if(0 == source) |
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1884 { |
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1885 /* Will return the number of sources halted */ |
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1886 return Internal_HaltChannel(-1, did_finish_naturally); |
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1887 } |
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1888 |
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1889 channel = Internal_GetChannel(source); |
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1890 if(-1 == channel) |
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1891 { |
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1892 ALmixer_SetError("Cannot halt source: %s", ALmixer_GetError()); |
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1893 return -1; |
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1894 } |
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1895 return Internal_HaltChannel(channel, did_finish_naturally); |
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1896 } |
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1897 |
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1898 |
0 | 1899 |
1900 /* Note: Behaves, almost like SDL_mixer, but keep in mind | |
1901 * that there is no "music" channel anymore, so 0 | |
1902 * will remove everything. (Note, I no longer allow 0 | |
1903 * so it gets set to the default number.) | |
1904 * Also, callbacks for deleted channels will not be called. | |
1905 * I really need to do error checking, for realloc and | |
1906 * GenSources, but reversing the damage is too painful | |
1907 * for me to think about at the moment, so it's not in here. | |
1908 */ | |
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1909 static ALint Internal_AllocateChannels(ALint numchans) |
0 | 1910 { |
1911 ALenum error; | |
1912 int i; | |
1913 /* Return info */ | |
1914 if(numchans < 0) | |
1915 { | |
1916 return Number_of_Channels_global; | |
1917 } | |
1918 if(0 == numchans) | |
1919 { | |
1920 numchans = ALMIXER_DEFAULT_NUM_CHANNELS; | |
1921 } | |
1922 /* No change */ | |
1923 if(numchans == Number_of_Channels_global) | |
1924 { | |
1925 return Number_of_Channels_global; | |
1926 } | |
1927 /* We need to increase the number of channels */ | |
1928 if(numchans > Number_of_Channels_global) | |
1929 { | |
1930 /* Not sure how safe this is, but SDL_mixer does it | |
1931 * the same way */ | |
1932 ALmixer_Channel_List = (struct ALmixer_Channel*) realloc( ALmixer_Channel_List, numchans * sizeof(struct ALmixer_Channel)); | |
1933 | |
1934 /* Allocate memory for the list of sources that map to the channels */ | |
1935 Source_Map_List = (Source_Map*) realloc(Source_Map_List, numchans * sizeof(Source_Map)); | |
1936 | |
1937 for(i=Number_of_Channels_global; i<numchans; i++) | |
1938 { | |
1939 Init_Channel(i); | |
1940 /* Generate a new source and associate it with the channel */ | |
1941 alGenSources(1, &ALmixer_Channel_List[i].alsource); | |
1942 if((error = alGetError()) != AL_NO_ERROR) | |
1943 { | |
1944 fprintf(stderr, "12Testing errpr before unqueue because getting stuff, for OS X this is expected: %s\n", | |
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1945 alGetString(error)); |
0 | 1946 } |
1947 /* Copy the source so the SourceMap has it too */ | |
1948 Source_Map_List[i].source = ALmixer_Channel_List[i].alsource; | |
1949 Source_Map_List[i].channel = i; | |
1950 /* Clean the channel because there are some things that need to | |
1951 * be done that can't happen until the source is set | |
1952 */ | |
1953 Clean_Channel(i); | |
1954 } | |
1955 | |
1956 /* The Source_Map_List must be sorted by source for binary searches | |
1957 */ | |
1958 qsort(Source_Map_List, numchans, sizeof(Source_Map), Compare_Source_Map); | |
1959 | |
1960 Number_of_Channels_global = numchans; | |
1961 return numchans; | |
1962 } | |
1963 /* Need to remove channels. This might be dangerous */ | |
1964 if(numchans < Number_of_Channels_global) | |
1965 { | |
1966 for(i=numchans; i<Number_of_Channels_global; i++) | |
1967 { | |
1968 /* Halt the channel */ | |
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1969 Internal_HaltChannel(i, AL_FALSE); |
0 | 1970 |
1971 /* Delete source associated with the channel */ | |
1972 alDeleteSources(1, &ALmixer_Channel_List[i].alsource); | |
1973 if((error = alGetError()) != AL_NO_ERROR) | |
1974 { | |
1975 fprintf(stderr, "13Testing error: %s\n", | |
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1976 alGetString(error)); |
0 | 1977 } |
1978 } | |
1979 | |
1980 | |
1981 /* Not sure how safe this is, but SDL_mixer does it | |
1982 * the same way */ | |
1983 ALmixer_Channel_List = (struct ALmixer_Channel*) realloc( ALmixer_Channel_List, numchans * sizeof(struct ALmixer_Channel)); | |
1984 | |
1985 /* The tricky part is that we must remove the entries | |
1986 * in the source map that correspond to the deleted channels. | |
1987 * We'll resort the map by channels so we can pick them off | |
1988 * in order. | |
1989 */ | |
1990 qsort(Source_Map_List, Number_of_Channels_global, sizeof(Source_Map), Compare_Source_Map_by_channel); | |
1991 | |
1992 /* Deallocate memory for the list of sources that map to the channels */ | |
1993 Source_Map_List = (Source_Map*) realloc(Source_Map_List, numchans * sizeof(Source_Map)); | |
1994 | |
1995 /* Now resort the map by source and the correct num of chans */ | |
1996 qsort(Source_Map_List, numchans, sizeof(Source_Map), Compare_Source_Map); | |
1997 | |
1998 /* Reset the number of channels */ | |
1999 Number_of_Channels_global = numchans; | |
2000 return numchans; | |
2001 } | |
2002 /* Shouldn't ever reach here */ | |
2003 return -1; | |
2004 | |
2005 } | |
2006 | |
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2007 static ALint Internal_ReserveChannels(ALint num) |
0 | 2008 { |
2009 /* Can't reserve more than the max num of channels */ | |
2010 /* Actually, I'll allow it for people who just want to | |
2011 * set the value really high to effectively disable | |
2012 * auto-assignment | |
2013 */ | |
2014 | |
2015 /* Return the current number of reserved channels */ | |
2016 if(num < 0) | |
2017 { | |
2018 return Number_of_Reserve_Channels_global; | |
2019 } | |
2020 Number_of_Reserve_Channels_global = num; | |
2021 return Number_of_Reserve_Channels_global; | |
2022 } | |
2023 | |
2024 | |
2025 /* This will rewind the SDL_Sound sample for streamed | |
2026 * samples and start buffering up the data for the next | |
2027 * playback. This may require samples to be halted | |
2028 */ | |
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2029 static ALint Internal_RewindData(ALmixer_Data* data) |
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2030 { |
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2031 ALint retval = 0; |
0 | 2032 /* |
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2033 ALint bytes_returned; |
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2034 ALint i; |
0 | 2035 */ |
2036 if(NULL == data) | |
2037 { | |
2038 ALmixer_SetError("Cannot rewind because data is NULL\n"); | |
2039 return -1; | |
2040 } | |
2041 | |
2042 | |
2043 /* Might have to require Halt */ | |
2044 /* Okay, we assume Halt or natural stop has already | |
2045 * cleared the data buffers | |
2046 */ | |
2047 if(data->in_use) | |
2048 { | |
2049 fprintf(stderr, "Warning sample is in use. May not be able to rewind\n"); | |
2050 /* | |
2051 ALmixer_SetError("Data is in use. Cannot rewind unless all sources using the data are halted\n"); | |
2052 return -1; | |
2053 */ | |
2054 } | |
2055 | |
2056 | |
2057 /* Because Seek can alter things even in predecoded data, | |
2058 * decoded data must also be rewound | |
2059 */ | |
2060 if(data->decoded_all) | |
2061 { | |
2062 data->eof = 0; | |
2063 | |
2064 #if 0 | |
2065 #if defined(DISABLE_PREDECODED_SEEK) | |
2066 /* Since we can't seek predecoded stuff, it should be rewound */ | |
2067 return 0; | |
2068 #elif !defined(DISABLE_SEEK_MEMORY_OPTIMIZATION) | |
2069 /* This case is if the Sound_Sample has been deleted. | |
2070 * It assumes the data is already at the beginning. | |
2071 */ | |
2072 if(NULL == data->sample) | |
2073 { | |
2074 return 0; | |
2075 } | |
2076 /* Else, the sample has already been reallocated, | |
2077 * and we can fall to normal behavior | |
2078 */ | |
2079 #endif | |
2080 #endif | |
2081 /* If access_data, was enabled, the sound sample | |
2082 * still exists and we can do stuff. | |
2083 * If it's NULL, we can't do anything, but | |
2084 * it should already be "rewound". | |
2085 */ | |
2086 if(NULL == data->sample) | |
2087 { | |
2088 return 0; | |
2089 } | |
2090 /* Else, the sample has already been reallocated, | |
2091 * and we can fall to normal behavior | |
2092 */ | |
2093 | |
2094 Set_Predecoded_Seek_Position(data, 0); | |
2095 /* | |
2096 return data->total_bytes; | |
2097 */ | |
2098 return 0; | |
2099 } | |
2100 | |
2101 /* Remaining stuff for streamed data */ | |
2102 | |
2103 fprintf(stderr, "Rewinding for stream\n"); | |
2104 data->eof = 0; | |
2105 retval = Sound_Rewind(data->sample); | |
2106 if(0 == retval) | |
2107 { | |
2108 ALmixer_SetError( Sound_GetError() ); | |
2109 return -1; | |
2110 } | |
2111 fprintf(stderr, "Rewinding succeeded\n"); | |
2112 fprintf(stderr, "calling GetMoreData for Rewinding for stream\n"); | |
2113 #if 0 | |
2114 /* Clear error */ | |
2115 alGetError(); | |
2116 for(i=0; i<data->num_buffers; i++) | |
2117 { | |
2118 bytes_returned = GetMoreData(data, data->buffer[i]); | |
2119 if(-1 == bytes_returned) | |
2120 { | |
2121 return -1; | |
2122 } | |
2123 else if(0 == bytes_returned) | |
2124 { | |
2125 return -1; | |
2126 } | |
2127 retval += bytes_returned; | |
2128 | |
2129 } | |
2130 #endif | |
2131 | |
2132 | |
2133 fprintf(stderr, "end Rewinding for stream\n"); | |
2134 | |
2135 return retval; | |
2136 } | |
2137 | |
2138 | |
2139 | |
2140 | |
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2141 static ALint Internal_RewindChannel(ALint channel) |
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2142 { |
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2143 ALint retval = 0; |
0 | 2144 ALenum error; |
2145 ALint state; | |
2146 | |
2147 if(channel >= Number_of_Channels_global) | |
2148 { | |
2149 ALmixer_SetError("Cannot rewind channel %d because it exceeds maximum number of channels (%d)\n", channel, Number_of_Channels_global); | |
2150 return -1; | |
2151 } | |
2152 | |
2153 if((error = alGetError()) != AL_NO_ERROR) | |
2154 { | |
2155 fprintf(stderr, "24Testing error: %s\n", | |
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2156 alGetString(error)); |
0 | 2157 } |
2158 /* Clear error */ | |
2159 alGetError(); | |
2160 | |
2161 /* If the user specified a specific channel */ | |
2162 if(channel >= 0) | |
2163 { | |
2164 /* only need to process channel if in use */ | |
2165 if(ALmixer_Channel_List[channel].channel_in_use) | |
2166 { | |
2167 | |
2168 /* What should I do? Do I just rewind the channel | |
2169 * or also rewind the data? Since the data is | |
2170 * shared, let's make it the user's responsibility | |
2171 * to rewind the data. | |
2172 */ | |
2173 if(ALmixer_Channel_List[channel].almixer_data->decoded_all) | |
2174 { | |
2175 alGetSourcei( | |
2176 ALmixer_Channel_List[channel].alsource, | |
2177 AL_SOURCE_STATE, &state | |
2178 ); | |
2179 if((error = alGetError()) != AL_NO_ERROR) | |
2180 { | |
2181 fprintf(stderr, "25Testing error: %s\n", | |
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2182 alGetString(error)); |
0 | 2183 } |
2184 alSourceRewind(ALmixer_Channel_List[channel].alsource); | |
2185 if((error = alGetError()) != AL_NO_ERROR) | |
2186 { | |
2187 ALmixer_SetError("%s", | |
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2188 alGetString(error) ); |
0 | 2189 retval = -1; |
2190 } | |
2191 /* Need to resume playback if it was originally playing */ | |
2192 if(AL_PLAYING == state) | |
2193 { | |
2194 alSourcePlay(ALmixer_Channel_List[channel].alsource); | |
2195 if((error = alGetError()) != AL_NO_ERROR) | |
2196 { | |
2197 ALmixer_SetError("%s", | |
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2198 alGetString(error) ); |
0 | 2199 retval = -1; |
2200 } | |
2201 } | |
2202 else if(AL_PAUSED == state) | |
2203 { | |
2204 /* HACK: The problem is that when paused, after | |
2205 * the Rewind, I can't get it off the INITIAL | |
2206 * state without restarting | |
2207 */ | |
2208 alSourcePlay(ALmixer_Channel_List[channel].alsource); | |
2209 if((error = alGetError()) != AL_NO_ERROR) | |
2210 { | |
2211 fprintf(stderr, "25Testing error: %s\n", | |
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2212 alGetString(error)); |
0 | 2213 } |
2214 alSourcePause(ALmixer_Channel_List[channel].alsource); | |
2215 if((error = alGetError()) != AL_NO_ERROR) | |
2216 { | |
2217 ALmixer_SetError("%s", | |
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2218 alGetString(error) ); |
0 | 2219 retval = -1; |
2220 } | |
2221 } | |
2222 } | |
2223 else | |
2224 { | |
2225 /* Streamed data is different. Rewinding the channel | |
2226 * does no good. Rewinding the data will have an | |
2227 * effect, but it will be lagged based on how | |
2228 * much data is queued. Recommend users call Halt | |
2229 * before rewind if they want immediate results. | |
2230 */ | |
2231 retval = Internal_RewindData(ALmixer_Channel_List[channel].almixer_data); | |
2232 } | |
2233 } | |
2234 } | |
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2235 /* The user wants to rewind all channels */ |
0 | 2236 else |
2237 { | |
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2238 ALint i; |
0 | 2239 for(i=0; i<Number_of_Channels_global; i++) |
2240 { | |
2241 fprintf(stderr, "in use %d\n", ALmixer_Channel_List[i].channel_in_use ); | |
2242 /* only need to process channel if in use */ | |
2243 if(ALmixer_Channel_List[i].channel_in_use) | |
2244 { | |
2245 /* What should I do? Do I just rewind the channel | |
2246 * or also rewind the data? Since the data is | |
2247 * shared, let's make it the user's responsibility | |
2248 * to rewind the data. | |
2249 */ | |
2250 if(ALmixer_Channel_List[i].almixer_data->decoded_all) | |
2251 { | |
2252 alGetSourcei( | |
2253 ALmixer_Channel_List[i].alsource, | |
2254 AL_SOURCE_STATE, &state | |
2255 ); | |
2256 if((error = alGetError()) != AL_NO_ERROR) | |
2257 { | |
2258 fprintf(stderr, "26Testing error: %s\n", | |
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2259 alGetString(error)); |
0 | 2260 } |
2261 alSourceRewind(ALmixer_Channel_List[i].alsource); | |
2262 if((error = alGetError()) != AL_NO_ERROR) | |
2263 { | |
2264 ALmixer_SetError("%s", | |
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2265 alGetString(error) ); |
0 | 2266 retval = -1; |
2267 } | |
2268 /* Need to resume playback if it was originally playing */ | |
2269 if(AL_PLAYING == state) | |
2270 { | |
2271 alSourcePlay(ALmixer_Channel_List[i].alsource); | |
2272 if((error = alGetError()) != AL_NO_ERROR) | |
2273 { | |
2274 ALmixer_SetError("%s", | |
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2275 alGetString(error) ); |
0 | 2276 retval = -1; |
2277 } | |
2278 } | |
2279 else if(AL_PAUSED == state) | |
2280 { | |
2281 /* HACK: The problem is that when paused, after | |
2282 * the Rewind, I can't get it off the INITIAL | |
2283 * state without restarting | |
2284 */ | |
2285 alSourcePlay(ALmixer_Channel_List[i].alsource); | |
2286 if((error = alGetError()) != AL_NO_ERROR) | |
2287 { | |
2288 fprintf(stderr, "27Testing error: %s\n", | |
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2289 alGetString(error)); |
0 | 2290 } |
2291 alSourcePause(ALmixer_Channel_List[i].alsource); | |
2292 if((error = alGetError()) != AL_NO_ERROR) | |
2293 { | |
2294 ALmixer_SetError("%s", | |
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2295 alGetString(error) ); |
0 | 2296 retval = -1; |
2297 } | |
2298 } | |
2299 } | |
2300 else | |
2301 { | |
2302 /* Streamed data is different. Rewinding the channel | |
2303 * does no good. Rewinding the data will have an | |
2304 * effect, but it will be lagged based on how | |
2305 * much data is queued. Recommend users call Halt | |
2306 * before rewind if they want immediate results. | |
2307 */ | |
2308 retval = Internal_RewindData(ALmixer_Channel_List[i].almixer_data); | |
2309 } | |
2310 } | |
2311 } | |
2312 } | |
2313 return retval; | |
2314 } | |
2315 | |
2316 | |
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|
2317 static ALint Internal_RewindSource(ALuint source) |
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|
2318 { |
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|
2319 ALint channel; |
0 | 2320 if(0 == source) |
2321 { | |
2
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|
2322 return Internal_RewindChannel(-1) + 1; |
0 | 2323 } |
2324 | |
2325 channel = Internal_GetChannel(source); | |
2326 if(-1 == channel) | |
2327 { | |
2328 ALmixer_SetError("Cannot rewind source: %s", ALmixer_GetError()); | |
2329 return 0; | |
2330 } | |
2
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|
2331 return Internal_RewindChannel(channel) + 1; |
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|
2332 } |
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|
2333 |
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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 static ALint Internal_PlayChannelTimed(ALint channel, ALmixer_Data* data, ALint loops, ALint ticks) |
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2339 { |
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|
2340 ALenum error; |
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|
2341 int ret_flag = 0; |
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2342 if(NULL == data) |
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|
2343 { |
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|
2344 ALmixer_SetError("Can't play because data is NULL\n"); |
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|
2345 return -1; |
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2346 } |
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2347 |
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2348 /* There isn't a good way to share streamed files because |
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2349 * the decoded data doesn't stick around. |
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2350 * You must "Load" a brand new instance of |
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|
2351 * the data. If you try using the same data, |
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2352 * bad things may happen. This check will attempt |
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2353 * to prevent sharing |
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|
2354 */ |
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|
2355 if(0 == data->decoded_all) |
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|
2356 { |
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2357 if(data->in_use) |
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2358 { |
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|
2359 ALmixer_SetError("Can't play shared streamed sample because it is already in use"); |
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2360 return -1; |
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|
2361 } |
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2362 |
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|
2363 /* Make sure SDL_sound sample is not at EOF. |
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2364 * This mainly affects streamed files, |
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2365 * so the check is placed here |
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|
2366 */ |
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|
2367 if(data->eof) |
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|
2368 { |
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2369 if( -1 == Internal_RewindData(data) ) |
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|
2370 { |
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2371 ALmixer_SetError("Can't play sample because it is at EOF and cannot rewind"); |
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|
2372 return -1; |
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|
2373 } |
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|
2374 } |
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|
2375 } |
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|
2376 /* We need to provide the user with the first available channel */ |
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|
2377 if(-1 == channel) |
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2378 { |
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|
2379 ALint i; |
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|
2380 for(i=Number_of_Reserve_Channels_global; i<Number_of_Channels_global; i++) |
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|
2381 { |
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|
2382 if(0 == ALmixer_Channel_List[i].channel_in_use) |
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|
2383 { |
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|
2384 channel = i; |
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|
2385 break; |
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|
2386 } |
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|
2387 } |
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2388 /* if we couldn't find a channel, return an error */ |
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|
2389 if(i == Number_of_Channels_global) |
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2390 { |
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|
2391 ALmixer_SetError("No channels available for playing"); |
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2392 return -1; |
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2393 } |
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|
2394 } |
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2395 /* If we didn't assign the channel number, make sure it's not |
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2396 * out of bounds or in use */ |
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2397 else |
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|
2398 { |
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2399 if(channel >= Number_of_Channels_global) |
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|
2400 { |
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|
2401 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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2402 return -1; |
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|
2403 } |
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2404 else if(ALmixer_Channel_List[channel].channel_in_use) |
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|
2405 { |
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|
2406 ALmixer_SetError("Requested channel (%d) is in use", channel, Number_of_Channels_global-1, Number_of_Channels_global); |
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2407 return -1; |
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2408 } |
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|
2409 } |
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2410 /* Make sure the user doesn't enter some meaningless value */ |
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2411 if(loops < -1) |
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2412 { |
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2413 loops = -1; |
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2414 } |
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2415 |
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2416 /* loops will probably have to change to be controlled by SDL_Sound */ |
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2417 |
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2418 /* Set up the initial values for playing */ |
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2419 ALmixer_Channel_List[channel].channel_in_use = 1; |
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2420 data->in_use++; |
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2421 |
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2422 /* Shouldn't need updating until a callback is fired |
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2423 * (assuming that we call Play in this function |
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2424 */ |
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2425 ALmixer_Channel_List[channel].needs_stream = 0; |
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2426 ALmixer_Channel_List[channel].almixer_data = data; |
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2427 ALmixer_Channel_List[channel].start_time = ALmixer_GetTicks(); |
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2428 |
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2429 /* If user entered -1 (or less), set to -1 */ |
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2430 if(ticks < 0) |
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2431 { |
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|
2432 ALmixer_Channel_List[channel].expire_ticks = -1; |
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2433 } |
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2434 else |
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2435 { |
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2436 ALmixer_Channel_List[channel].expire_ticks = ticks; |
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2437 } |
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2438 |
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2439 |
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2440 ALmixer_Channel_List[channel].halted = 0; |
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2441 ALmixer_Channel_List[channel].paused = 0; |
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2442 |
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2443 /* Ran just use OpenAL to control loops if predecoded and infinite */ |
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2444 ALmixer_Channel_List[channel].loops = loops; |
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2445 if( (-1 == loops) && (data->decoded_all) ) |
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|
2446 { |
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|
2447 alSourcei(ALmixer_Channel_List[channel].alsource, AL_LOOPING, AL_TRUE); |
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|
2448 } |
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diff
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|
2449 else |
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1
diff
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|
2450 { |
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diff
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|
2451 alSourcei(ALmixer_Channel_List[channel].alsource, AL_LOOPING, AL_FALSE); |
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diff
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2452 } |
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1
diff
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|
2453 if((error = alGetError()) != AL_NO_ERROR) |
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diff
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|
2454 { |
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1
diff
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|
2455 fprintf(stderr, "13Testing error: %s\n", |
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diff
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|
2456 alGetString(error)); |
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diff
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|
2457 } |
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diff
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|
2458 |
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diff
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|
2459 #if 0 |
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diff
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|
2460 /* Because of the corner case, predecoded |
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diff
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|
2461 * files must add +1 to the loops. |
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diff
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|
2462 * Streams do not have this problem |
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diff
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|
2463 * because they can use the eof flag to |
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diff
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|
2464 * avoid the conflict. |
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1
diff
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|
2465 * Sharing data chunks prevents the use of the eof flag. |
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diff
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|
2466 * Since streams, cannot share, only predecoded |
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|
2467 * files are affected |
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diff
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|
2468 */ |
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diff
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|
2469 if(data->decoded_all) |
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|
2470 { |
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|
2471 /* Corner Case: Now that play calls are pushed |
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|
2472 * off to update(), the start call must |
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diff
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|
2473 * also come through here. So, start loops |
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diff
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|
2474 * must be +1 |
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diff
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|
2475 */ |
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diff
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|
2476 if(-1 == loops) |
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|
2477 { |
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diff
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|
2478 /* -1 is a special case, and you don't want |
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diff
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|
2479 * to add +1 to it */ |
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diff
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|
2480 ALmixer_Channel_List[channel].loops = -1; |
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diff
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|
2481 alSourcei(ALmixer_Channel_List[channel].alsource, AL_LOOPING, AL_TRUE); |
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diff
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|
2482 } |
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diff
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|
2483 else |
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diff
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|
2484 { |
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diff
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|
2485 ALmixer_Channel_List[channel].loops = loops+1; |
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diff
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|
2486 alSourcei(ALmixer_Channel_List[channel].alsource, AL_LOOPING, AL_FALSE); |
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diff
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|
2487 } |
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diff
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|
2488 } |
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|
2489 else |
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diff
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|
2490 { |
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diff
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|
2491 ALmixer_Channel_List[channel].loops = loops; |
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diff
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|
2492 /* Can we really loop on streamed data? */ |
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diff
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|
2493 alSourcei(ALmixer_Channel_List[channel].alsource, AL_LOOPING, AL_TRUE); |
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diff
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|
2494 } |
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1
diff
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|
2495 #endif |
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diff
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|
2496 |
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diff
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|
2497 /* Should I start playing here or pass the buck to update? */ |
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1
diff
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|
2498 /* Unlike SDL_SoundMixer, I think I'll do it here because |
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1
diff
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|
2499 * this library isn't a *total* hack and OpenAL has more |
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diff
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|
2500 * built in functionality I need, so less needs to be |
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diff
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|
2501 * controlled and directed through the update function. |
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diff
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|
2502 * The downside is less functionality is centralized. |
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diff
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|
2503 * The upside is that the update function should be |
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diff
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|
2504 * easier to maintain. |
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diff
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|
2505 */ |
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diff
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|
2506 |
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diff
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|
2507 /* Clear the error flag */ |
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diff
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|
2508 alGetError(); |
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diff
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|
2509 if(data->decoded_all) |
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diff
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|
2510 { |
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diff
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|
2511 /* Bind the data to the source */ |
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diff
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|
2512 alSourcei( |
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1
diff
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|
2513 ALmixer_Channel_List[channel].alsource, |
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1
diff
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|
2514 AL_BUFFER, |
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diff
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|
2515 data->buffer[0]); |
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diff
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|
2516 if((error = alGetError()) != AL_NO_ERROR) |
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1
diff
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|
2517 { |
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1
diff
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|
2518 ALmixer_SetError("Could not bind data to source: %s", |
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1
diff
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|
2519 alGetString(error) ); |
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diff
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|
2520 Clean_Channel(channel); |
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diff
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|
2521 return -1; |
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1
diff
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|
2522 } |
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diff
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|
2523 |
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diff
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|
2524 /* Make data available if access_data is enabled */ |
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1
diff
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|
2525 Invoke_Predecoded_Channel_Data_Callback(channel, data); |
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1
diff
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|
2526 } |
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1
diff
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|
2527 else |
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1
diff
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|
2528 { |
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diff
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|
2529 /* Need to use the streaming buffer for binding */ |
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diff
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|
2530 |
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diff
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|
2531 ALuint bytes_returned; |
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diff
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|
2532 ALuint j; |
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1
diff
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|
2533 data->num_buffers_in_use=0; |
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diff
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|
2534 /****** MODIFICATION must go here *********/ |
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diff
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|
2535 /* Since buffer queuing is pushed off until here to |
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diff
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|
2536 * avoid buffer conflicts, we must start reading |
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diff
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|
2537 * data here. First we make sure we have at least one |
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diff
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|
2538 * packet. Then we queue up until we hit our limit. |
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diff
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|
2539 */ |
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diff
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|
2540 bytes_returned = GetMoreData( |
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diff
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|
2541 data, |
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diff
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|
2542 data->buffer[0]); |
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|
2543 if(0 == bytes_returned) |
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|
2544 { |
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diff
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|
2545 /* No data or error */ |
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|
2546 ALmixer_SetError("Could not get data for streamed PlayChannel: %s", ALmixer_GetError()); |
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diff
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|
2547 Clean_Channel(channel); |
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diff
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|
2548 return -1; |
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1
diff
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|
2549 } |
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diff
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|
2550 /* Increment the number of buffers in use */ |
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diff
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|
2551 data->num_buffers_in_use++; |
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|
2552 |
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|
2553 |
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diff
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|
2554 /* Now we need to fill up the rest of the buffers. |
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diff
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|
2555 * There is a corner case where we run out of data |
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|
2556 * before the last buffer is filled. |
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|
2557 * Stop conditions are we run out of |
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|
2558 * data or we max out our preload buffers. |
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diff
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|
2559 */ |
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|
2560 |
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|
2561 fprintf(stderr, "Filling buffer #%d (AL id is %d)\n", 0, data->buffer[0]); |
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|
2562 for(j=1; j<data->num_startup_buffers; j++) |
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|
2563 { |
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diff
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|
2564 fprintf(stderr, "Filling buffer #%d (AL id is %d)\n", j, data->buffer[j]); |
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diff
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|
2565 /* |
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1
diff
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|
2566 fprintf(stderr, ">>>>>>>>>>>>>>>>>>HACK for GetMoreData2\n"); |
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1
diff
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|
2567 */ |
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1
diff
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|
2568 bytes_returned = GetMoreData( |
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1
diff
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|
2569 data, |
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|
2570 data->buffer[j]); |
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|
2571 /* |
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|
2572 * This might be a problem. I made a mistake with the types. I accidentally |
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|
2573 * made the bytes returned an ALint and returned -1 on error. |
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|
2574 * Bytes returned should be a ALuint, so now I no longer have a -1 case |
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2575 * to check. I hope I didn't break anything here |
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|
2576 */ |
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|
2577 #if 0 |
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|
2578 if(bytes_returned < 0) |
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|
2579 { |
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|
2580 /* Error found */ |
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|
2581 ALmixer_SetError("Could not get data for additional startup buffers for PlayChannel: %s", ALmixer_GetError()); |
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2582 /* We'll continue on because we do have some valid data */ |
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2583 ret_flag = -1; |
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2584 break; |
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2585 } |
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2586 else if(0 == bytes_returned) |
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2587 #endif |
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2588 if(0 == bytes_returned) |
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2589 { |
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2590 /* No more data to buffer */ |
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2591 /* Check for loops */ |
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|
2592 if( ALmixer_Channel_List[channel].loops != 0 ) |
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2593 { |
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|
2594 fprintf(stderr, "Need to rewind. In RAMPUP, handling loop\n"); |
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|
2595 if(0 == Sound_Rewind(data->sample)) |
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2596 { |
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2597 fprintf(stderr, "error in rewind\n"); |
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2598 ALmixer_SetError( Sound_GetError() ); |
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2599 ALmixer_Channel_List[channel].loops = 0; |
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|
2600 ret_flag = -1; |
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|
2601 /* We'll continue on because we do have some valid data */ |
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2602 break; |
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2603 } |
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2604 /* Remember to reset the data->eof flag */ |
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2605 data->eof = 0; |
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|
2606 if(ALmixer_Channel_List[channel].loops > 0) |
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2607 { |
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2608 ALmixer_Channel_List[channel].loops--; |
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|
2609 fprintf(stderr, "Inside 000 >>>>>>>>>>Loops=%d\n", ALmixer_Channel_List[channel].loops); |
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2610 } |
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2611 /* Would like to redo the loop, but due to |
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2612 * Sound_Rewind() bugs, we would risk falling |
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2613 * into an infinite loop |
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|
2614 */ |
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2615 bytes_returned = GetMoreData( |
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|
2616 data, |
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|
2617 data->buffer[j]); |
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2618 if(bytes_returned <= 0) |
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2619 { |
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2620 ALmixer_SetError("Could not get data: %s", ALmixer_GetError()); |
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2621 /* We'll continue on because we do have some valid data */ |
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2622 ret_flag = -1; |
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2623 break; |
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2624 } |
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2625 } |
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2626 else |
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2627 { |
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2628 /* No loops to do so quit here */ |
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2629 break; |
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2630 } |
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2631 } |
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2632 /* Increment the number of buffers in use */ |
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2633 data->num_buffers_in_use++; |
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2634 } |
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2635 /* |
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2636 fprintf(stderr, "In PlayChannel, about to queue: source=%d, num_buffers_in_use=%d\n", |
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2637 ALmixer_Channel_List[channel].alsource, |
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2638 data->num_buffers_in_use); |
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2639 */ |
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2640 |
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2641 alSourceQueueBuffers( |
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2642 ALmixer_Channel_List[channel].alsource, |
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2643 data->num_buffers_in_use, |
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2644 data->buffer); |
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2645 if((error = alGetError()) != AL_NO_ERROR) |
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2646 { |
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2647 ALmixer_SetError("Could not bind data to source: %s", |
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2648 alGetString(error) ); |
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2649 Clean_Channel(channel); |
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2650 return -1; |
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2651 } |
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2652 /* This is part of the hideous Nvidia workaround. In order to figure out |
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2653 * which buffer to show during callbacks (for things like |
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2654 * o-scopes), I must keep a copy of the buffers that are queued in my own |
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2655 * data structure. This code will be called only if |
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2656 * "access_data" was set, indicated by whether the queue is NULL. |
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2657 */ |
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2658 if(data->circular_buffer_queue != NULL) |
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2659 { |
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2660 ALuint k; |
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2661 ALuint queue_ret_flag; |
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2662 for(k=0; k<data->num_buffers_in_use; k++) |
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2663 { |
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2664 // fprintf(stderr, "56c: CircularQueue_PushBack.\n"); |
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2665 queue_ret_flag = CircularQueueUnsignedInt_PushBack(data->circular_buffer_queue, data->buffer[k]); |
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2666 if(0 == queue_ret_flag) |
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2667 { |
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2668 fprintf(stderr, "Serious internal error: CircularQueue could not push into queue.\n"); |
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2669 ALmixer_SetError("Serious internal error: CircularQueue failed to push into queue"); |
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2670 } |
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2671 /* |
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2672 else |
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2673 { |
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2674 fprintf(stderr, "Queue in PlayTimed\n"); |
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2675 CircularQueueUnsignedInt_Print(data->circular_buffer_queue); |
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2676 } |
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2677 */ |
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2678 } |
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2679 } |
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2680 |
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2681 |
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2682 /****** END **********/ |
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2683 } |
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2684 /* We have finished loading the data (predecoded or queued) |
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2685 * so now we can play |
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2686 */ |
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2687 alSourcePlay(ALmixer_Channel_List[channel].alsource); |
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2688 if((error = alGetError()) != AL_NO_ERROR) |
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2689 { |
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2690 ALmixer_SetError("Play failed: %s", |
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2691 alGetString(error) ); |
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2692 Clean_Channel(channel); |
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2693 return -1; |
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2694 } |
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2695 |
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2696 /* Add to the counter that something is playing */ |
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2697 Is_Playing_global++; |
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2698 if(-1 == ret_flag) |
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2699 { |
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2700 fprintf(stderr, "BACKDOOR ERROR >>>>>>>>>>>>>>>>>>\n"); |
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2701 return -1; |
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2702 } |
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2703 return channel; |
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2704 } |
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2705 |
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2706 |
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2707 /* In case the user wants to specify a source instead of a channel, |
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2708 * they may use this function. This function will look up the |
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2709 * source-to-channel map, and convert the call into a |
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2710 * PlayChannelTimed() function call. |
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2711 * Returns the channel it's being played on. |
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2712 * Note: If you are prefer this method, then you need to be careful |
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2713 * about using PlayChannel, particularly if you request the |
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2714 * first available channels because source and channels have |
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2715 * a one-to-one mapping in this API. It is quite easy for |
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2716 * a channel/source to already be in use because of this. |
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2717 * In this event, an error message will be returned to you. |
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2718 */ |
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2719 static ALuint Internal_PlaySourceTimed(ALuint source, ALmixer_Data* data, ALint loops, ALint ticks) |
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2720 { |
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2721 ALint channel; |
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2722 ALint retval; |
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2723 if(0 == source) |
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2724 { |
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2725 retval = Internal_PlayChannelTimed(-1, data, loops, ticks); |
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2726 if(-1 == retval) |
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2727 { |
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2728 return 0; |
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2729 } |
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2730 else |
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2731 { |
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2732 return Internal_GetSource(retval); |
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2733 } |
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2734 } |
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2735 |
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2736 channel = Internal_GetChannel(source); |
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2737 if(-1 == channel) |
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2738 { |
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2739 ALmixer_SetError("Cannot Play source: %s", ALmixer_GetError()); |
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2740 return 0; |
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2741 } |
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2742 retval = Internal_PlayChannelTimed(channel, data, loops, ticks); |
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2743 if(-1 == retval) |
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2744 { |
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2745 return 0; |
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2746 } |
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2747 else |
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2748 { |
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2749 return source; |
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2750 } |
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2751 /* make compiler happy */ |
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2752 return 0; |
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2753 } |
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2754 |
0 | 2755 |
2756 | |
2757 | |
2758 /* Returns the channel or number of channels actually paused */ | |
2759 | |
2
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2760 static ALint Internal_PauseChannel(ALint channel) |
0 | 2761 { |
2762 ALenum error; | |
2763 ALint state; | |
2
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2764 ALint retval = 0; |
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2765 ALint counter = 0; |
0 | 2766 |
2767 if(channel >= Number_of_Channels_global) | |
2768 { | |
2769 ALmixer_SetError("Cannot pause channel %d because it exceeds maximum number of channels (%d)\n", channel, Number_of_Channels_global); | |
2770 return -1; | |
2771 } | |
2772 | |
2773 if((error = alGetError()) != AL_NO_ERROR) | |
2774 { | |
2775 fprintf(stderr, "28Testing error: %s\n", | |
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2776 alGetString(error)); |
0 | 2777 } |
2778 /* Clear error */ | |
2779 alGetError(); | |
2780 | |
2781 /* If the user specified a specific channel */ | |
2782 if(channel >= 0) | |
2783 { | |
2784 fprintf(stderr, "Pause on channel %d\n", channel); | |
2785 /* only need to process channel if in use */ | |
2786 if(ALmixer_Channel_List[channel].channel_in_use) | |
2787 { | |
2788 /* We don't want to repause if already | |
2789 * paused because the fadeout/expire | |
2790 * timing will get messed up | |
2791 */ | |
2792 alGetSourcei( | |
2793 ALmixer_Channel_List[channel].alsource, | |
2794 AL_SOURCE_STATE, &state | |
2795 ); | |
2796 if((error = alGetError()) != AL_NO_ERROR) | |
2797 { | |
2798 fprintf(stderr, "29Testing error: %s\n", | |
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2799 alGetString(error)); |
0 | 2800 } |
2801 if(AL_PLAYING == state) | |
2802 { | |
2803 /* Count the actual number of channels being paused */ | |
2804 counter++; | |
2805 | |
2806 alSourcePause(ALmixer_Channel_List[channel].alsource); | |
2807 if((error = alGetError()) != AL_NO_ERROR) | |
2808 { | |
2809 ALmixer_SetError("%s", | |
2
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2810 alGetString(error) ); |
0 | 2811 retval = -1; |
2812 } | |
2813 /* We need to pause the expire time count down */ | |
2814 if(ALmixer_Channel_List[channel].expire_ticks != -1) | |
2815 { | |
2
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2816 ALuint current_time = ALmixer_GetTicks(); |
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2817 ALuint diff_time; |
0 | 2818 diff_time = current_time - |
2819 ALmixer_Channel_List[channel].start_time; | |
2820 /* When we unpause, we will want to reset | |
2821 * the start time so we can continue | |
2822 * to base calculations off GetTicks(). | |
2823 * This means we need to subtract the amount | |
2824 * of time already used up from expire_ticks. | |
2825 */ | |
2826 ALmixer_Channel_List[channel].expire_ticks = | |
2827 ALmixer_Channel_List[channel].expire_ticks - | |
2828 diff_time; | |
2829 /* Because -1 is a special value, we can't | |
2830 * allow the time to go negative | |
2831 */ | |
2832 if(ALmixer_Channel_List[channel].expire_ticks < 0) | |
2833 { | |
2834 ALmixer_Channel_List[channel].expire_ticks = 0; | |
2835 } | |
2836 } | |
2837 /* Do the same as expire time for fading */ | |
2838 if(ALmixer_Channel_List[channel].fade_enabled) | |
2839 { | |
2
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2840 ALuint current_time = ALmixer_GetTicks(); |
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2841 ALuint diff_time; |
0 | 2842 diff_time = current_time - |
2843 ALmixer_Channel_List[channel].fade_start_time; | |
2844 /* When we unpause, we will want to reset | |
2845 * the start time so we can continue | |
2846 * to base calculations off GetTicks(). | |
2847 * This means we need to subtract the amount | |
2848 * of time already used up from expire_ticks. | |
2849 */ | |
2850 ALmixer_Channel_List[channel].fade_expire_ticks = | |
2851 ALmixer_Channel_List[channel].fade_expire_ticks - | |
2852 diff_time; | |
2853 /* Don't allow the time to go negative */ | |
2854 if(ALmixer_Channel_List[channel].expire_ticks < 0) | |
2855 { | |
2856 ALmixer_Channel_List[channel].expire_ticks = 0; | |
2857 } | |
2858 } /* End fade check */ | |
2859 } /* End if PLAYING */ | |
2860 } /* End If in use */ | |
2861 } /* End specific channel */ | |
2862 /* The user wants to halt all channels */ | |
2863 else | |
2864 { | |
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2865 ALint i; |
0 | 2866 for(i=0; i<Number_of_Channels_global; i++) |
2867 { | |
2868 fprintf(stderr, "Pausing channel %d\n", i); | |
2869 fprintf(stderr, "in use %d\n", ALmixer_Channel_List[i].channel_in_use ); | |
2870 /* only need to process channel if in use */ | |
2871 if(ALmixer_Channel_List[i].channel_in_use) | |
2872 { | |
2873 /* We don't want to repause if already | |
2874 * paused because the fadeout/expire | |
2875 * timing will get messed up | |
2876 */ | |
2877 alGetSourcei( | |
2878 ALmixer_Channel_List[i].alsource, | |
2879 AL_SOURCE_STATE, &state | |
2880 ); | |
2881 if((error = alGetError()) != AL_NO_ERROR) | |
2882 { | |
2883 fprintf(stderr, "30Testing error: %s\n", | |
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2884 alGetString(error)); |
0 | 2885 } |
2886 if(AL_PLAYING == state) | |
2887 { | |
2888 /* Count the actual number of channels being paused */ | |
2889 counter++; | |
2890 | |
2891 fprintf(stderr, "SourcePause %d\n", i); | |
2892 alSourcePause(ALmixer_Channel_List[i].alsource); | |
2893 if((error = alGetError()) != AL_NO_ERROR) | |
2894 { | |
2895 ALmixer_SetError("%s", | |
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2896 alGetString(error) ); |
0 | 2897 retval = -1; |
2898 } | |
2899 /* We need to pause the expire time count down */ | |
2900 if(ALmixer_Channel_List[i].expire_ticks != -1) | |
2901 { | |
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2902 ALuint current_time = ALmixer_GetTicks(); |
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2903 ALuint diff_time; |
0 | 2904 diff_time = current_time - |
2905 ALmixer_Channel_List[i].start_time; | |
2906 /* When we unpause, we will want to reset | |
2907 * the start time so we can continue | |
2908 * to base calculations off GetTicks(). | |
2909 * This means we need to subtract the amount | |
2910 * of time already used up from expire_ticks. | |
2911 */ | |
2912 ALmixer_Channel_List[i].expire_ticks = | |
2913 ALmixer_Channel_List[i].expire_ticks - | |
2914 diff_time; | |
2915 /* Because -1 is a special value, we can't | |
2916 * allow the time to go negative | |
2917 */ | |
2918 if(ALmixer_Channel_List[i].expire_ticks < 0) | |
2919 { | |
2920 ALmixer_Channel_List[i].expire_ticks = 0; | |
2921 } | |
2922 } | |
2923 /* Do the same as expire time for fading */ | |
2924 if(ALmixer_Channel_List[i].fade_enabled) | |
2925 { | |
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2926 ALuint current_time = ALmixer_GetTicks(); |
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2927 ALuint diff_time; |
0 | 2928 diff_time = current_time - |
2929 ALmixer_Channel_List[i].fade_start_time; | |
2930 /* When we unpause, we will want to reset | |
2931 * the start time so we can continue | |
2932 * to base calculations off GetTicks(). | |
2933 * This means we need to subtract the amount | |
2934 * of time already used up from expire_ticks. | |
2935 */ | |
2936 ALmixer_Channel_List[i].fade_expire_ticks = | |
2937 ALmixer_Channel_List[i].fade_expire_ticks - | |
2938 diff_time; | |
2939 /* Don't allow the time to go negative */ | |
2940 if(ALmixer_Channel_List[i].expire_ticks < 0) | |
2941 { | |
2942 ALmixer_Channel_List[i].expire_ticks = 0; | |
2943 } | |
2944 } /* End fade check */ | |
2945 } /* End if PLAYING */ | |
2946 } /* End channel in use */ | |
2947 } /* End for-loop */ | |
2948 } | |
2949 if(-1 == retval) | |
2950 { | |
2951 return -1; | |
2952 } | |
2953 return counter; | |
2954 } | |
2955 | |
2956 /* Returns the channel or number of channels actually paused */ | |
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2957 static ALint Internal_PauseSource(ALuint source) |
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2958 { |
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2959 ALint channel; |
0 | 2960 if(0 == source) |
2961 { | |
2962 return Internal_PauseChannel(-1); | |
2963 } | |
2964 | |
2965 channel = Internal_GetChannel(source); | |
2966 if(-1 == channel) | |
2967 { | |
2968 ALmixer_SetError("Cannot pause source: %s", ALmixer_GetError()); | |
2969 return -1; | |
2970 } | |
2971 return Internal_PauseChannel(channel); | |
2972 } | |
2973 | |
2974 | |
2975 | |
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2976 static ALint Internal_ResumeChannel(ALint channel) |
0 | 2977 { |
2978 ALint state; | |
2979 ALenum error; | |
2
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2980 ALint retval = 0; |
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2981 ALint counter = 0; |
0 | 2982 |
2983 if(channel >= Number_of_Channels_global) | |
2984 { | |
2985 ALmixer_SetError("Cannot pause channel %d because it exceeds maximum number of channels (%d)\n", channel, Number_of_Channels_global); | |
2986 return -1; | |
2987 } | |
2988 | |
2989 if((error = alGetError()) != AL_NO_ERROR) | |
2990 { | |
2991 fprintf(stderr, "31Testing error: %s\n", | |
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2992 alGetString(error)); |
0 | 2993 } |
2994 /* Clear error */ | |
2995 alGetError(); | |
2996 | |
2997 /* If the user specified a specific channel */ | |
2998 if(channel >= 0) | |
2999 { | |
3000 fprintf(stderr, "Pause on channel %d\n", channel); | |
3001 /* only need to process channel if in use */ | |
3002 if(ALmixer_Channel_List[channel].channel_in_use) | |
3003 { | |
3004 alGetSourcei( | |
3005 ALmixer_Channel_List[channel].alsource, | |
3006 AL_SOURCE_STATE, &state | |
3007 ); | |
3008 if((error = alGetError()) != AL_NO_ERROR) | |
3009 { | |
3010 fprintf(stderr, "32Testing error: %s\n", | |
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3011 alGetString(error)); |
0 | 3012 } |
3013 if(AL_PAUSED == state) | |
3014 { | |
3015 /* Count the actual number of channels resumed */ | |
3016 counter++; | |
3017 | |
3018 /* We need to resume the expire time count down */ | |
3019 if(ALmixer_Channel_List[channel].expire_ticks != -1) | |
3020 { | |
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3021 ALmixer_Channel_List[channel].start_time = ALmixer_GetTicks(); |
0 | 3022 } |
3023 /* Do the same as expire time for fading */ | |
3024 if(ALmixer_Channel_List[channel].fade_enabled) | |
3025 { | |
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3026 ALmixer_Channel_List[channel].fade_start_time = ALmixer_GetTicks(); |
0 | 3027 } |
3028 | |
3029 alSourcePlay(ALmixer_Channel_List[channel].alsource); | |
3030 if((error = alGetError()) != AL_NO_ERROR) | |
3031 { | |
3032 ALmixer_SetError("%s", | |
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3033 alGetString(error) ); |
0 | 3034 retval = -1; |
3035 } | |
3036 } | |
3037 fprintf(stderr, "Pause on channel %d, channel in use\n", channel); | |
3038 } | |
3039 } | |
3040 /* The user wants to halt all channels */ | |
3041 else | |
3042 { | |
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3043 ALint i; |
0 | 3044 for(i=0; i<Number_of_Channels_global; i++) |
3045 { | |
3046 fprintf(stderr, "Pausing channel %d\n", i); | |
3047 fprintf(stderr, "in use %d\n", ALmixer_Channel_List[i].channel_in_use ); | |
3048 /* only need to process channel if in use */ | |
3049 if(ALmixer_Channel_List[i].channel_in_use) | |
3050 { | |
3051 fprintf(stderr, "SourcePause %d\n", i); | |
3052 alGetSourcei( | |
3053 ALmixer_Channel_List[i].alsource, | |
3054 AL_SOURCE_STATE, &state | |
3055 ); | |
3056 if((error = alGetError()) != AL_NO_ERROR) | |
3057 { | |
3058 fprintf(stderr, "33Testing error: %s\n", | |
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3059 alGetString(error)); |
0 | 3060 } |
3061 if(AL_PAUSED == state) | |
3062 { | |
3063 /* Count the actual number of channels resumed */ | |
3064 counter++; | |
3065 | |
3066 /* We need to resume the expire time count down */ | |
3067 if(ALmixer_Channel_List[i].expire_ticks != -1) | |
3068 { | |
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3069 ALmixer_Channel_List[i].start_time = ALmixer_GetTicks(); |
0 | 3070 } |
3071 /* Do the same as expire time for fading */ | |
3072 if(ALmixer_Channel_List[i].fade_enabled) | |
3073 { | |
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3074 ALmixer_Channel_List[i].fade_start_time = ALmixer_GetTicks(); |
0 | 3075 } |
3076 | |
3077 alSourcePlay(ALmixer_Channel_List[i].alsource); | |
3078 if((error = alGetError()) != AL_NO_ERROR) | |
3079 { | |
3080 ALmixer_SetError("%s", | |
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3081 alGetString(error) ); |
0 | 3082 retval = -1; |
3083 } | |
3084 } | |
3085 } | |
3086 } | |
3087 } | |
3088 if(-1 == retval) | |
3089 { | |
3090 return -1; | |
3091 } | |
3092 return counter; | |
3093 } | |
3094 | |
3095 | |
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3096 static ALint Internal_ResumeSource(ALuint source) |
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3097 { |
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3098 ALint channel; |
0 | 3099 if(0 == source) |
3100 { | |
3101 return Internal_ResumeChannel(-1); | |
3102 } | |
3103 | |
3104 channel = Internal_GetChannel(source); | |
3105 if(-1 == channel) | |
3106 { | |
3107 ALmixer_SetError("Cannot resume source: %s", ALmixer_GetError()); | |
3108 return -1; | |
3109 } | |
3110 return Internal_ResumeChannel(channel); | |
3111 } | |
3112 | |
3113 | |
3114 /* Might consider setting eof to 0 as a "feature" | |
3115 * This will allow seek to end to stay there because | |
3116 * Play automatically rewinds if at the end */ | |
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3117 static ALint Internal_SeekData(ALmixer_Data* data, ALuint msec) |
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3118 { |
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3119 ALint retval; |
0 | 3120 |
3121 if(NULL == data) | |
3122 { | |
3123 ALmixer_SetError("Cannot Seek because data is NULL"); | |
3124 return -1; | |
3125 } | |
3126 | |
3127 /* Seek for predecoded files involves moving the chunk pointer around */ | |
3128 if(data->decoded_all) | |
3129 { | |
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3130 ALuint byte_position; |
0 | 3131 |
3132 /* OpenAL doesn't seem to like it if I change the buffer | |
3133 * while playing (crashes), so I must require that Seek only | |
3134 * be done when the data is not in use. | |
3135 * Since data may be shared among multiple sources, | |
3136 * I can't shut them down myself, so I have to return an error. | |
3137 */ | |
3138 if(data->in_use) | |
3139 { | |
3140 ALmixer_SetError("Cannot seek on predecoded data while instances are playing"); | |
3141 return -1; | |
3142 } | |
3143 #if 0 | |
3144 #if defined(DISABLE_PREDECODED_SEEK) | |
3145 ALmixer_SetError("Seek support for predecoded samples was not compiled in"); | |
3146 return -1; | |
3147 | |
3148 #elif !defined(DISABLE_SEEK_MEMORY_OPTIMIZATION) | |
3149 /* By default, ALmixer frees the Sound_Sample for predecoded | |
3150 * samples because of the potential memory waste. | |
3151 * However, to seek a sample, we need to have a full | |
3152 * copy of the data around. So the strategy is to | |
3153 * recreate a hackish Sound_Sample to be used for seeking | |
3154 * purposes. If Sound_Sample is NULL, we will reallocate | |
3155 * memory for it and then procede as if everything | |
3156 * was normal. | |
3157 */ | |
3158 if(NULL == data->sample) | |
3159 { | |
3160 if( -1 == Reconstruct_Sound_Sample(data) ) | |
3161 { | |
3162 return -1; | |
3163 } | |
3164 } | |
3165 #endif | |
3166 #endif | |
3167 /* If access_data was set, then we still have the | |
3168 * Sound_Sample and we can move around in the data. | |
3169 * If it was not set, the data has been freed and we | |
3170 * cannot do anything because there is no way to | |
3171 * recover the data because OpenAL won't let us | |
3172 * get access to the buffers | |
3173 */ | |
3174 if(NULL == data->sample) | |
3175 { | |
3176 ALmixer_SetError("Cannot seek because access_data flag was set false when data was initialized"); | |
3177 return -1; | |
3178 } | |
3179 | |
3180 fprintf(stderr, "Calling convert\n"); | |
3181 byte_position = Convert_Msec_To_Byte_Pos(&data->sample->desired, msec); | |
3182 fprintf(stderr, "Calling Set_Predecoded_Seek...%d\n", byte_position); | |
3183 return( Set_Predecoded_Seek_Position(data, byte_position) ); | |
3184 } | |
3185 else | |
3186 { | |
3187 /* Reset eof flag?? */ | |
3188 data->eof = 0; | |
3189 retval = Sound_Seek(data->sample, msec); | |
3190 if(0 == retval) | |
3191 { | |
3192 ALmixer_SetError(Sound_GetError()); | |
3193 | |
3194 fprintf(stderr, "Sound seek error: %s\n", ALmixer_GetError()); | |
3195 /* Try rewinding to clean up? */ | |
3196 /* | |
3197 Internal_RewindData(data); | |
3198 */ | |
3199 return -1; | |
3200 } | |
3201 return 0; | |
3202 } | |
3203 | |
3204 return 0; | |
3205 } | |
3206 | |
3207 | |
3208 | |
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3209 static ALint Internal_FadeInChannelTimed(ALint channel, ALmixer_Data* data, ALint loops, ALuint fade_ticks, ALint expire_ticks) |
0 | 3210 { |
3211 ALfloat value; | |
3212 ALenum error; | |
3213 ALfloat original_value; | |
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3214 ALuint current_time = ALmixer_GetTicks(); |
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3215 ALint retval; |
0 | 3216 |
3217 | |
3218 | |
3219 if(channel >= Number_of_Channels_global) | |
3220 { | |
3221 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); | |
3222 return -1; | |
3223 } | |
3224 /* Let's call PlayChannelTimed to do the job. | |
3225 * There are two catches: | |
3226 * First is that we must set the volumes before the play call(s). | |
3227 * Second is that we must initialize the channel values | |
3228 */ | |
3229 | |
3230 if(channel < 0) | |
3231 { | |
3232 /* This might cause a problem for threads/race conditions. | |
3233 * We need to set the volume on an unknown channel, | |
3234 * so we need to request a channel first. Remember | |
3235 * that requesting a channel doesn't lock and it | |
3236 * could be surrendered to somebody else before we claim it. | |
3237 */ | |
3238 channel = Internal_GetChannel(0); | |
3239 if(-1 == channel) | |
3240 { | |
3241 return -1; | |
3242 } | |
3243 } | |
3244 else if(ALmixer_Channel_List[channel].channel_in_use) | |
3245 { | |
3246 ALmixer_SetError("Channel %d is already in use", channel); | |
3247 return -1; | |
3248 } | |
3249 | |
3250 | |
3251 /* Get the original volume in case of a problem */ | |
3252 alGetSourcef(ALmixer_Channel_List[channel].alsource, | |
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3253 AL_GAIN, &original_value); |
0 | 3254 |
3255 if((error = alGetError()) != AL_NO_ERROR) | |
3256 { | |
3257 fprintf(stderr, "35Testing error: %s\n", | |
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3258 alGetString(error)); |
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3259 } |
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3260 ALmixer_Channel_List[channel].fade_end_volume = original_value; |
0 | 3261 |
3262 /* Get the Min volume */ | |
3263 alGetSourcef(ALmixer_Channel_List[channel].alsource, | |
3264 AL_MIN_GAIN, &value); | |
3265 if((error = alGetError()) != AL_NO_ERROR) | |
3266 { | |
3267 fprintf(stderr, "36Testing error: %s\n", | |
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3268 alGetString(error)); |
0 | 3269 } |
3270 ALmixer_Channel_List[channel].fade_start_volume = value; | |
3271 fprintf(stderr, "MIN gain: %f\n", value); | |
3272 | |
3273 /* Set the actual volume */ | |
3274 alSourcef(ALmixer_Channel_List[channel].alsource, | |
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3275 AL_GAIN, value); |
0 | 3276 if((error = alGetError()) != AL_NO_ERROR) |
3277 { | |
3278 fprintf(stderr, "37Testing error: %s\n", | |
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3279 alGetString(error)); |
0 | 3280 } |
3281 | |
3282 | |
3283 /* Now call PlayChannelTimed */ | |
3284 retval = Internal_PlayChannelTimed(channel, data, loops, expire_ticks); | |
3285 if(-1 == retval) | |
3286 { | |
3287 /* Chance of failure is actually pretty high since | |
3288 * a channel might already be in use or streamed | |
3289 * data can be shared | |
3290 */ | |
3291 /* Restore the original value to avoid accidental | |
3292 * distruption of playback | |
3293 */ | |
3294 alSourcef(ALmixer_Channel_List[channel].alsource, | |
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3295 AL_GAIN, original_value); |
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3296 if((error = alGetError()) != AL_NO_ERROR) |
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3297 { |
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3298 fprintf(stderr, "38Testing error: %s\n", |
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3299 alGetString(error)); |
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3300 } |
0 | 3301 return retval; |
3302 } | |
3303 | |
3304 /* We can't accept 0 as a value because of div-by-zero. | |
3305 * If zero, just call PlayChannelTimed at normal | |
3306 * volume | |
3307 */ | |
3308 if(0 == fade_ticks) | |
3309 { | |
3310 alSourcef(ALmixer_Channel_List[channel].alsource, | |
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3311 AL_GAIN, |
0 | 3312 ALmixer_Channel_List[channel].fade_end_volume |
3313 ); | |
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3314 if((error = alGetError()) != AL_NO_ERROR) |
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3315 { |
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3316 fprintf(stderr, "39Testing error: %s\n", |
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3317 alGetString(error)); |
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3318 } |
0 | 3319 |
3320 return retval; | |
3321 } | |
3322 | |
3323 /* Enable fading effects via the flag */ | |
3324 ALmixer_Channel_List[channel].fade_enabled = 1; | |
3325 /* Set fade start time */ | |
3326 ALmixer_Channel_List[channel].fade_start_time | |
3327 = ALmixer_Channel_List[channel].start_time; | |
3328 fprintf(stderr, "Current time =%d\n", current_time); | |
3329 /* Set the fade expire ticks */ | |
3330 ALmixer_Channel_List[channel].fade_expire_ticks = fade_ticks; | |
3331 | |
3332 /* Set 1/(endtime-starttime) or 1/deltaT */ | |
3333 ALmixer_Channel_List[channel].fade_inv_time = 1.0f / fade_ticks; | |
3334 | |
3335 return retval; | |
3336 | |
3337 } | |
3338 | |
3339 | |
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3340 static ALuint Internal_FadeInSourceTimed(ALuint source, ALmixer_Data* data, ALint loops, ALuint fade_ticks, ALint expire_ticks) |
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3341 { |
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3342 ALint channel; |
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3343 ALint retval; |
0 | 3344 if(0 == source) |
3345 { | |
3346 retval = Internal_FadeInChannelTimed(-1, data, loops, fade_ticks, expire_ticks); | |
3347 if(-1 == retval) | |
3348 { | |
3349 return 0; | |
3350 } | |
3351 else | |
3352 { | |
3353 return Internal_GetSource(retval); | |
3354 } | |
3355 } | |
3356 | |
3357 channel = Internal_GetChannel(source); | |
3358 if(-1 == channel) | |
3359 { | |
3360 ALmixer_SetError("Cannot FadeIn source: %s", ALmixer_GetError()); | |
3361 return 0; | |
3362 } | |
3363 retval = Internal_FadeInChannelTimed(channel, data, loops, fade_ticks, expire_ticks); | |
3364 if(-1 == retval) | |
3365 { | |
3366 return 0; | |
3367 } | |
3368 else | |
3369 { | |
3370 return source; | |
3371 } | |
3372 /* make compiler happy */ | |
3373 return 0; | |
3374 } | |
3375 | |
3376 | |
3377 | |
3378 | |
3379 /* Will fade out currently playing channels. | |
3380 * It starts at the current volume level and goes down */ | |
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3381 static ALint Internal_FadeOutChannel(ALint channel, ALuint ticks) |
0 | 3382 { |
3383 ALfloat value; | |
3384 ALenum error; | |
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3385 ALuint current_time = ALmixer_GetTicks(); |
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|
3386 ALuint counter = 0; |
0 | 3387 |
3388 /* We can't accept 0 as a value because of div-by-zero. | |
3389 * If zero, just call Halt at normal | |
3390 * volume | |
3391 */ | |
3392 if(0 == ticks) | |
3393 { | |
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3394 return Internal_HaltChannel(channel, AL_TRUE); |
0 | 3395 } |
3396 | |
3397 | |
3398 if(channel >= Number_of_Channels_global) | |
3399 { | |
3400 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); | |
3401 return -1; | |
3402 } | |
3403 | |
3404 if(channel >= 0) | |
3405 { | |
3406 if(ALmixer_Channel_List[channel].channel_in_use) | |
3407 { | |
3408 /* Get the current volume */ | |
3409 alGetSourcef(ALmixer_Channel_List[channel].alsource, | |
2
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|
3410 AL_GAIN, &value); |
0 | 3411 ALmixer_Channel_List[channel].fade_start_volume = value; |
3412 if((error = alGetError()) != AL_NO_ERROR) | |
3413 { | |
3414 fprintf(stderr, "40Testing error: %s\n", | |
2
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|
3415 alGetString(error)); |
0 | 3416 } |
3417 | |
3418 /* Get the Min volume */ | |
3419 alGetSourcef(ALmixer_Channel_List[channel].alsource, | |
3420 AL_MIN_GAIN, &value); | |
3421 if((error = alGetError()) != AL_NO_ERROR) | |
3422 { | |
3423 fprintf(stderr, "41Testing error: %s\n", | |
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|
3424 alGetString(error)); |
0 | 3425 } |
3426 ALmixer_Channel_List[channel].fade_end_volume = value; | |
3427 fprintf(stderr, "MIN gain: %f\n", value); | |
3428 | |
3429 /* Set expire start time */ | |
3430 ALmixer_Channel_List[channel].start_time = current_time; | |
3431 /* Set the expire ticks */ | |
3432 ALmixer_Channel_List[channel].expire_ticks = ticks; | |
3433 /* Set fade start time */ | |
3434 ALmixer_Channel_List[channel].fade_start_time = current_time; | |
3435 /* Set the fade expire ticks */ | |
3436 ALmixer_Channel_List[channel].fade_expire_ticks = ticks; | |
3437 /* Enable fading effects via the flag */ | |
3438 ALmixer_Channel_List[channel].fade_enabled = 1; | |
3439 | |
3440 /* Set 1/(endtime-starttime) or 1/deltaT */ | |
3441 ALmixer_Channel_List[channel].fade_inv_time = 1.0f / ticks; | |
3442 | |
3443 counter++; | |
3444 } | |
3445 } | |
3446 /* Else need to fade out all channels */ | |
3447 else | |
3448 { | |
2
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|
3449 ALint i; |
0 | 3450 for(i=0; i<Number_of_Channels_global; i++) |
3451 { | |
3452 if(ALmixer_Channel_List[i].channel_in_use) | |
3453 { | |
3454 /* Get the current volume */ | |
3455 alGetSourcef(ALmixer_Channel_List[i].alsource, | |
2
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|
3456 AL_GAIN, &value); |
0 | 3457 ALmixer_Channel_List[i].fade_start_volume = value; |
3458 if((error = alGetError()) != AL_NO_ERROR) | |
3459 { | |
3460 fprintf(stderr, "42Testing error: %s\n", | |
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|
3461 alGetString(error)); |
0 | 3462 } |
3463 | |
3464 /* Get the Min volume */ | |
3465 alGetSourcef(ALmixer_Channel_List[i].alsource, | |
3466 AL_MIN_GAIN, &value); | |
3467 if((error = alGetError()) != AL_NO_ERROR) | |
3468 { | |
3469 fprintf(stderr, "43Testing error: %s\n", | |
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3470 alGetString(error)); |
0 | 3471 } |
3472 ALmixer_Channel_List[i].fade_end_volume = value; | |
3473 fprintf(stderr, "MIN gain: %f\n", value); | |
3474 | |
3475 /* Set expire start time */ | |
3476 ALmixer_Channel_List[i].start_time = current_time; | |
3477 /* Set the expire ticks */ | |
3478 ALmixer_Channel_List[i].expire_ticks = ticks; | |
3479 /* Set fade start time */ | |
3480 ALmixer_Channel_List[i].fade_start_time = current_time; | |
3481 /* Set the fade expire ticks */ | |
3482 ALmixer_Channel_List[i].fade_expire_ticks = ticks; | |
3483 /* Enable fading effects via the flag */ | |
3484 ALmixer_Channel_List[i].fade_enabled = 1; | |
3485 | |
3486 /* Set 1/(endtime-starttime) or 1/deltaT */ | |
3487 ALmixer_Channel_List[i].fade_inv_time = 1.0f / ticks; | |
3488 | |
3489 counter++; | |
3490 } | |
3491 } /* End for loop */ | |
3492 } | |
3493 return counter; | |
3494 } | |
3495 | |
3496 | |
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|
3497 static ALint Internal_FadeOutSource(ALuint source, ALuint ticks) |
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3498 { |
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|
3499 ALint channel; |
0 | 3500 if(0 == source) |
3501 { | |
3502 return Internal_FadeOutChannel(-1, ticks); | |
3503 } | |
3504 | |
3505 channel = Internal_GetChannel(source); | |
3506 if(-1 == channel) | |
3507 { | |
3508 ALmixer_SetError("Cannot FadeOut source: %s", ALmixer_GetError()); | |
3509 return -1; | |
3510 } | |
3511 return Internal_FadeOutChannel(channel, ticks); | |
3512 } | |
3513 | |
3514 | |
3515 /* Will fade currently playing channels. | |
3516 * It starts at the current volume level and go to target | |
3517 * Only affects channels that are playing | |
3518 */ | |
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|
3519 static ALint Internal_FadeChannel(ALint channel, ALuint ticks, ALfloat volume) |
0 | 3520 { |
3521 ALfloat value; | |
3522 ALenum error; | |
2
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3523 ALuint current_time = ALmixer_GetTicks(); |
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3524 ALuint counter = 0; |
0 | 3525 |
3526 if(channel >= Number_of_Channels_global) | |
3527 { | |
3528 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); | |
3529 return -1; | |
3530 } | |
3531 | |
3532 if(channel >= 0) | |
3533 { | |
3534 if(volume < ALmixer_Channel_List[channel].min_volume) | |
3535 { | |
3536 volume = ALmixer_Channel_List[channel].min_volume; | |
3537 } | |
3538 else if(volume > ALmixer_Channel_List[channel].max_volume) | |
3539 { | |
3540 volume = ALmixer_Channel_List[channel].max_volume; | |
3541 } | |
3542 | |
3543 if(ALmixer_Channel_List[channel].channel_in_use) | |
3544 { | |
3545 if(ticks > 0) | |
3546 { | |
3547 /* Get the current volume */ | |
3548 alGetSourcef(ALmixer_Channel_List[channel].alsource, | |
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|
3549 AL_GAIN, &value); |
0 | 3550 if((error = alGetError()) != AL_NO_ERROR) |
3551 { | |
3552 fprintf(stderr, "44Testing error: %s\n", | |
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|
3553 alGetString(error)); |
0 | 3554 } |
3555 ALmixer_Channel_List[channel].fade_start_volume = value; | |
3556 | |
3557 /* Set the target volume */ | |
3558 ALmixer_Channel_List[channel].fade_end_volume = volume; | |
3559 | |
3560 /* Set fade start time */ | |
3561 ALmixer_Channel_List[channel].fade_start_time = current_time; | |
3562 /* Set the fade expire ticks */ | |
3563 ALmixer_Channel_List[channel].fade_expire_ticks = ticks; | |
3564 /* Enable fading effects via the flag */ | |
3565 ALmixer_Channel_List[channel].fade_enabled = 1; | |
3566 | |
3567 /* Set 1/(endtime-starttime) or 1/deltaT */ | |
3568 ALmixer_Channel_List[channel].fade_inv_time = 1.0f / ticks; | |
3569 } | |
3570 else | |
3571 { | |
3572 alSourcef(ALmixer_Channel_List[channel].alsource, | |
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|
3573 AL_GAIN, volume); |
0 | 3574 if((error = alGetError()) != AL_NO_ERROR) |
3575 { | |
3576 fprintf(stderr, "45Testing error: %s\n", | |
2
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|
3577 alGetString(error)); |
0 | 3578 } |
3579 } | |
3580 counter++; | |
3581 } | |
3582 } | |
3583 /* Else need to fade out all channels */ | |
3584 else | |
3585 { | |
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3586 ALint i; |
0 | 3587 for(i=0; i<Number_of_Channels_global; i++) |
3588 { | |
3589 if(volume < ALmixer_Channel_List[i].min_volume) | |
3590 { | |
3591 volume = ALmixer_Channel_List[i].min_volume; | |
3592 } | |
3593 else if(volume > ALmixer_Channel_List[i].max_volume) | |
3594 { | |
3595 volume = ALmixer_Channel_List[i].max_volume; | |
3596 } | |
3597 | |
3598 if(ALmixer_Channel_List[i].channel_in_use) | |
3599 { | |
3600 if(ticks > 0) | |
3601 { | |
3602 /* Get the current volume */ | |
3603 alGetSourcef(ALmixer_Channel_List[i].alsource, | |
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|
3604 AL_GAIN, &value); |
0 | 3605 if((error = alGetError()) != AL_NO_ERROR) |
3606 { | |
3607 fprintf(stderr, "46Testing error: %s\n", | |
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|
3608 alGetString(error)); |
0 | 3609 } |
3610 ALmixer_Channel_List[i].fade_start_volume = value; | |
3611 | |
3612 /* Set target volume */ | |
3613 ALmixer_Channel_List[i].fade_end_volume = volume; | |
3614 | |
3615 /* Set fade start time */ | |
3616 ALmixer_Channel_List[i].fade_start_time = current_time; | |
3617 /* Set the fade expire ticks */ | |
3618 ALmixer_Channel_List[i].fade_expire_ticks = ticks; | |
3619 /* Enable fading effects via the flag */ | |
3620 ALmixer_Channel_List[i].fade_enabled = 1; | |
3621 | |
3622 /* Set 1/(endtime-starttime) or 1/deltaT */ | |
3623 ALmixer_Channel_List[i].fade_inv_time = 1.0f / ticks; | |
3624 } | |
3625 else | |
3626 { | |
3627 alSourcef(ALmixer_Channel_List[i].alsource, | |
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|
3628 AL_GAIN, volume); |
0 | 3629 if((error = alGetError()) != AL_NO_ERROR) |
3630 { | |
3631 fprintf(stderr, "47Testing error: %s\n", | |
2
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|
3632 alGetString(error)); |
0 | 3633 } |
3634 } | |
3635 counter++; | |
3636 } | |
3637 } /* End for loop */ | |
3638 } | |
3639 return counter; | |
3640 } | |
3641 | |
2
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3642 static ALint Internal_FadeSource(ALuint source, ALuint ticks, ALfloat volume) |
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3643 { |
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|
3644 ALint channel; |
0 | 3645 if(0 == source) |
3646 { | |
3647 return Internal_FadeChannel(-1, ticks, volume); | |
3648 } | |
3649 | |
3650 channel = Internal_GetChannel(source); | |
3651 if(-1 == channel) | |
3652 { | |
3653 ALmixer_SetError("Cannot Fade source: %s", ALmixer_GetError()); | |
3654 return -1; | |
3655 } | |
3656 return Internal_FadeChannel(channel, ticks, volume); | |
3657 } | |
3658 | |
3659 | |
3660 | |
3661 | |
3662 /* Set a volume regardless if it's in use or not. | |
3663 */ | |
2
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|
3664 static ALboolean Internal_SetVolumeChannel(ALint channel, ALfloat volume) |
0 | 3665 { |
3666 ALenum error; | |
2
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|
3667 ALboolean retval = AL_TRUE; |
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|
3668 |
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|
3669 if(channel >= Number_of_Channels_global) |
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|
3670 { |
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|
3671 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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|
3672 return AL_FALSE; |
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|
3673 } |
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|
3674 |
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|
3675 if(channel >= 0) |
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|
3676 { |
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|
3677 if(volume < 0.0f) |
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|
3678 { |
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|
3679 volume = 0.0f; |
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|
3680 } |
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|
3681 else if(volume > 1.0f) |
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|
3682 { |
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|
3683 volume = 1.0f; |
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|
3684 } |
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|
3685 alSourcef(ALmixer_Channel_List[channel].alsource, |
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|
3686 AL_GAIN, volume); |
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3687 if((error = alGetError()) != AL_NO_ERROR) |
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|
3688 { |
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|
3689 ALmixer_SetError("%s", |
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|
3690 alGetString(error) ); |
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|
3691 retval = AL_FALSE; |
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|
3692 } |
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|
3693 } |
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|
3694 else |
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|
3695 { |
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|
3696 ALint i; |
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|
3697 for(i=0; i<Number_of_Channels_global; i++) |
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|
3698 { |
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|
3699 if(volume < 0.0f) |
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|
3700 { |
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|
3701 volume = 0.0f; |
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|
3702 } |
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|
3703 else if(volume > 1.0f) |
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|
3704 { |
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|
3705 volume = 1.0f; |
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|
3706 } |
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|
3707 alSourcef(ALmixer_Channel_List[i].alsource, |
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|
3708 AL_GAIN, volume); |
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changeset
|
3709 if((error = alGetError()) != AL_NO_ERROR) |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3710 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3711 ALmixer_SetError("%s", |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3712 alGetString(error) ); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3713 retval = AL_FALSE; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3714 } |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3715 } |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3716 } |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3717 return retval; |
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 |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3720 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
|
3721 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3722 ALint channel; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3723 if(0 == source) |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3724 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3725 return Internal_SetVolumeChannel(-1, volume); |
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 |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3728 channel = Internal_GetChannel(source); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3729 if(-1 == channel) |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3730 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3731 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
|
3732 return AL_FALSE; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3733 } |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3734 return Internal_SetVolumeChannel(channel, volume); |
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 |
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 static ALfloat Internal_GetVolumeChannel(ALint channel) |
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 ALfloat value; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3741 ALenum error; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3742 ALfloat running_total = 0.0f; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3743 ALfloat retval = 0.0f; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3744 |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3745 if(channel >= Number_of_Channels_global) |
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 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
|
3748 return -1.0f; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3749 } |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3750 |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3751 if(channel >= 0) |
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 alGetSourcef(ALmixer_Channel_List[channel].alsource, |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3754 AL_GAIN, &value); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3755 if((error = alGetError()) != AL_NO_ERROR) |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3756 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3757 ALmixer_SetError("%s", alGetString(error) ); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3758 retval = -1.0f; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3759 } |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3760 else |
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 retval = value; |
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 } |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3765 else |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3766 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3767 ALint i; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3768 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
|
3769 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3770 alGetSourcef(ALmixer_Channel_List[i].alsource, |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3771 AL_GAIN, &value); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3772 if((error = alGetError()) != AL_NO_ERROR) |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3773 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3774 ALmixer_SetError("%s", alGetString(error) ); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3775 retval = -1; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3776 } |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3777 else |
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 running_total += value; |
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 } |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3782 if(0 == Number_of_Channels_global) |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3783 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3784 ALmixer_SetError("No channels are allocated"); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3785 retval = -1.0f; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3786 } |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3787 else |
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 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
|
3790 } |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3791 } |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3792 return retval; |
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 |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3795 static ALfloat Internal_GetVolumeSource(ALuint source) |
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 ALint channel; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3798 if(0 == source) |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3799 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3800 return Internal_GetVolumeChannel(-1); |
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 |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3803 channel = Internal_GetChannel(source); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3804 if(-1 == channel) |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3805 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3806 ALmixer_SetError("Cannot GetVolume: %s", ALmixer_GetError()); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3807 return -1.0f; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3808 } |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3809 |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3810 return Internal_GetVolumeChannel(channel); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3811 } |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3812 |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3813 |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3814 |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3815 /* Set a volume regardless if it's in use or not. |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3816 */ |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3817 static ALboolean Internal_SetMaxVolumeChannel(ALint channel, ALfloat volume) |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3818 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3819 ALenum error; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3820 ALboolean retval = AL_TRUE; |
0 | 3821 |
3822 if(channel >= Number_of_Channels_global) | |
3823 { | |
3824 ALmixer_SetError("Requested channel (%d) exceeds maximum channel (%d) because only %d channels are allocated", channel, Number_of_Channels_global-1, Number_of_Channels_global); | |
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3825 return AL_FALSE; |
0 | 3826 } |
3827 | |
3828 if(channel >= 0) | |
3829 { | |
3830 if(volume < 0.0f) | |
3831 { | |
3832 volume = 0.0f; | |
3833 } | |
3834 else if(volume > 1.0f) | |
3835 { | |
3836 volume = 1.0f; | |
3837 } | |
3838 ALmixer_Channel_List[channel].max_volume = volume; | |
3839 alSourcef(ALmixer_Channel_List[channel].alsource, | |
3840 AL_MAX_GAIN, volume); | |
3841 if((error = alGetError()) != AL_NO_ERROR) | |
3842 { | |
3843 ALmixer_SetError("%s", | |
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3844 alGetString(error) ); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3845 retval = AL_FALSE; |
0 | 3846 } |
3847 if(ALmixer_Channel_List[channel].max_volume < ALmixer_Channel_List[channel].min_volume) | |
3848 { | |
3849 ALmixer_Channel_List[channel].min_volume = volume; | |
3850 alSourcef(ALmixer_Channel_List[channel].alsource, | |
3851 AL_MIN_GAIN, volume); | |
3852 if((error = alGetError()) != AL_NO_ERROR) | |
3853 { | |
3854 ALmixer_SetError("%s", | |
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3855 alGetString(error) ); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3856 retval = AL_FALSE; |
0 | 3857 } |
3858 } | |
3859 } | |
3860 else | |
3861 { | |
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3862 ALint i; |
0 | 3863 for(i=0; i<Number_of_Channels_global; i++) |
3864 { | |
3865 if(volume < 0.0f) | |
3866 { | |
3867 volume = 0.0f; | |
3868 } | |
3869 else if(volume > 1.0f) | |
3870 { | |
3871 volume = 1.0f; | |
3872 } | |
3873 ALmixer_Channel_List[i].max_volume = volume; | |
3874 alSourcef(ALmixer_Channel_List[i].alsource, | |
3875 AL_MAX_GAIN, volume); | |
3876 if((error = alGetError()) != AL_NO_ERROR) | |
3877 { | |
3878 ALmixer_SetError("%s", | |
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3879 alGetString(error) ); |
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3880 retval = AL_FALSE; |
0 | 3881 } |
3882 if(ALmixer_Channel_List[i].max_volume < ALmixer_Channel_List[i].min_volume) | |
3883 { | |
3884 ALmixer_Channel_List[i].min_volume = volume; | |
3885 alSourcef(ALmixer_Channel_List[i].alsource, | |
3886 AL_MIN_GAIN, volume); | |
3887 if((error = alGetError()) != AL_NO_ERROR) | |
3888 { | |
3889 ALmixer_SetError("%s", | |
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3890 alGetString(error) ); |
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3891 retval = AL_FALSE; |
0 | 3892 } |
3893 } | |
3894 } | |
3895 } | |
3896 return retval; | |
3897 } | |
3898 | |
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3899 static ALint Internal_SetMaxVolumeSource(ALuint source, ALfloat volume) |
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3900 { |
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3901 ALint channel; |
0 | 3902 if(0 == source) |
3903 { | |
3904 return Internal_SetMaxVolumeChannel(-1, volume); | |
3905 } | |
3906 | |
3907 channel = Internal_GetChannel(source); | |
3908 if(-1 == channel) | |
3909 { | |
3910 ALmixer_SetError("Cannot SetMaxVolume: %s", ALmixer_GetError()); | |
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3911 return AL_FALSE; |
0 | 3912 } |
3913 return Internal_SetMaxVolumeChannel(channel, volume); | |
3914 } | |
3915 | |
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3916 static ALfloat Internal_GetMaxVolumeChannel(ALint channel) |
0 | 3917 { |
3918 /* | |
3919 ALfloat value; | |
3920 ALenum error; | |
3921 */ | |
3922 ALfloat running_total = 0.0f; | |
3923 ALfloat retval = 0.0f; | |
3924 | |
3925 if(channel >= Number_of_Channels_global) | |
3926 { | |
3927 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); | |
3928 return -1.0f; | |
3929 } | |
3930 | |
3931 if(channel >= 0) | |
3932 { | |
3933 /* | |
3934 alGetSourcef(ALmixer_Channel_List[channel].alsource, | |
3935 AL_GAIN, &value); | |
3936 if((error = alGetError()) != AL_NO_ERROR) | |
3937 { | |
3938 ALmixer_SetError("%s", | |
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3939 alGetString(error) ); |
0 | 3940 retval = -1.0f; |
3941 } | |
3942 else | |
3943 { | |
3944 retval = value; | |
3945 } | |
3946 */ | |
3947 retval = ALmixer_Channel_List[channel].max_volume; | |
3948 | |
3949 } | |
3950 else | |
3951 { | |
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3952 ALint i; |
0 | 3953 for(i=0; i<Number_of_Channels_global; i++) |
3954 { | |
3955 /* | |
3956 alGetSourcef(ALmixer_Channel_List[i].alsource, | |
3957 AL_GAIN, &value); | |
3958 if((error = alGetError()) != AL_NO_ERROR) | |
3959 { | |
3960 ALmixer_SetError("%s", | |
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3961 alGetString(error) ); |
0 | 3962 retval = -1; |
3963 } | |
3964 else | |
3965 { | |
3966 running_total += value; | |
3967 } | |
3968 */ | |
3969 running_total += ALmixer_Channel_List[i].max_volume; | |
3970 } | |
3971 if(0 == Number_of_Channels_global) | |
3972 { | |
3973 ALmixer_SetError("No channels are allocated"); | |
3974 retval = -1.0f; | |
3975 } | |
3976 else | |
3977 { | |
3978 retval = running_total / Number_of_Channels_global; | |
3979 } | |
3980 } | |
3981 return retval; | |
3982 } | |
3983 | |
3984 static ALfloat Internal_GetMaxVolumeSource(ALuint source) | |
3985 { | |
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3986 ALint channel; |
0 | 3987 if(0 == source) |
3988 { | |
3989 return Internal_GetMaxVolumeChannel(-1); | |
3990 } | |
3991 | |
3992 channel = Internal_GetChannel(source); | |
3993 if(-1 == channel) | |
3994 { | |
3995 ALmixer_SetError("Cannot GetVolume: %s", ALmixer_GetError()); | |
3996 return -1.0f; | |
3997 } | |
3998 | |
3999 return Internal_GetMaxVolumeChannel(channel); | |
4000 } | |
4001 | |
4002 | |
4003 /* Set a volume regardless if it's in use or not. | |
4004 */ | |
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4005 static ALboolean Internal_SetMinVolumeChannel(ALint channel, ALfloat volume) |
0 | 4006 { |
4007 ALenum error; | |
2
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4008 ALboolean retval = AL_TRUE; |
0 | 4009 |
4010 if(channel >= Number_of_Channels_global) | |
4011 { | |
4012 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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4013 return AL_FALSE; |
0 | 4014 } |
4015 | |
4016 if(channel >= 0) | |
4017 { | |
4018 if(volume < 0.0f) | |
4019 { | |
4020 volume = 0.0f; | |
4021 } | |
4022 else if(volume > 1.0f) | |
4023 { | |
4024 volume = 1.0f; | |
4025 } | |
4026 ALmixer_Channel_List[channel].min_volume = volume; | |
4027 alSourcef(ALmixer_Channel_List[channel].alsource, | |
4028 AL_MIN_GAIN, volume); | |
4029 if((error = alGetError()) != AL_NO_ERROR) | |
4030 { | |
4031 ALmixer_SetError("%s", | |
2
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4032 alGetString(error) ); |
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4033 retval = AL_FALSE; |
0 | 4034 } |
4035 if(ALmixer_Channel_List[channel].max_volume < ALmixer_Channel_List[channel].min_volume) | |
4036 { | |
4037 ALmixer_Channel_List[channel].max_volume = volume; | |
4038 alSourcef(ALmixer_Channel_List[channel].alsource, | |
4039 AL_MAX_GAIN, volume); | |
4040 if((error = alGetError()) != AL_NO_ERROR) | |
4041 { | |
4042 ALmixer_SetError("%s", | |
2
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4043 alGetString(error) ); |
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4044 retval = AL_FALSE; |
0 | 4045 } |
4046 } | |
4047 } | |
4048 else | |
4049 { | |
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4050 ALint i; |
0 | 4051 for(i=0; i<Number_of_Channels_global; i++) |
4052 { | |
4053 if(volume < 0.0f) | |
4054 { | |
4055 volume = 0.0f; | |
4056 } | |
4057 else if(volume > 1.0f) | |
4058 { | |
4059 volume = 1.0f; | |
4060 } | |
4061 ALmixer_Channel_List[i].min_volume = volume; | |
4062 alSourcef(ALmixer_Channel_List[i].alsource, | |
4063 AL_MIN_GAIN, volume); | |
4064 if((error = alGetError()) != AL_NO_ERROR) | |
4065 { | |
4066 ALmixer_SetError("%s", | |
2
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4067 alGetString(error) ); |
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4068 retval = AL_FALSE; |
0 | 4069 } |
4070 if(ALmixer_Channel_List[i].max_volume < ALmixer_Channel_List[i].min_volume) | |
4071 { | |
4072 ALmixer_Channel_List[i].max_volume = volume; | |
4073 alSourcef(ALmixer_Channel_List[i].alsource, | |
4074 AL_MAX_GAIN, volume); | |
4075 if((error = alGetError()) != AL_NO_ERROR) | |
4076 { | |
4077 ALmixer_SetError("%s", | |
2
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4078 alGetString(error) ); |
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4079 retval = AL_FALSE; |
0 | 4080 } |
4081 } | |
4082 } | |
4083 } | |
4084 return retval; | |
4085 } | |
4086 | |
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4087 static ALboolean Internal_SetMinVolumeSource(ALuint source, ALfloat volume) |
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4088 { |
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4089 ALint channel; |
0 | 4090 if(0 == source) |
4091 { | |
4092 return Internal_SetMinVolumeChannel(-1, volume); | |
4093 } | |
4094 | |
4095 channel = Internal_GetChannel(source); | |
4096 if(-1 == channel) | |
4097 { | |
4098 ALmixer_SetError("Cannot SetMaxVolume: %s", ALmixer_GetError()); | |
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4099 return AL_FALSE; |
0 | 4100 } |
4101 return Internal_SetMinVolumeChannel(channel, volume); | |
4102 } | |
4103 | |
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4104 static ALfloat Internal_GetMinVolumeChannel(ALint channel) |
0 | 4105 { |
4106 /* | |
4107 ALfloat value; | |
4108 ALenum error; | |
4109 */ | |
4110 ALfloat running_total = 0.0f; | |
4111 ALfloat retval = 0.0f; | |
4112 | |
4113 if(channel >= Number_of_Channels_global) | |
4114 { | |
4115 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); | |
4116 return -1.0f; | |
4117 } | |
4118 | |
4119 if(channel >= 0) | |
4120 { | |
4121 /* | |
4122 alGetSourcef(ALmixer_Channel_List[channel].alsource, | |
4123 AL_GAIN, &value); | |
4124 if((error = alGetError()) != AL_NO_ERROR) | |
4125 { | |
4126 ALmixer_SetError("%s", | |
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4127 alGetString(error) ); |
0 | 4128 retval = -1.0f; |
4129 } | |
4130 else | |
4131 { | |
4132 retval = value; | |
4133 } | |
4134 */ | |
4135 retval = ALmixer_Channel_List[channel].min_volume; | |
4136 | |
4137 } | |
4138 else | |
4139 { | |
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4140 ALint i; |
0 | 4141 for(i=0; i<Number_of_Channels_global; i++) |
4142 { | |
4143 /* | |
4144 alGetSourcef(ALmixer_Channel_List[i].alsource, | |
4145 AL_GAIN, &value); | |
4146 if((error = alGetError()) != AL_NO_ERROR) | |
4147 { | |
4148 ALmixer_SetError("%s", | |
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4149 alGetString(error) ); |
0 | 4150 retval = -1; |
4151 } | |
4152 else | |
4153 { | |
4154 running_total += value; | |
4155 } | |
4156 */ | |
4157 running_total += ALmixer_Channel_List[i].min_volume; | |
4158 } | |
4159 if(0 == Number_of_Channels_global) | |
4160 { | |
4161 ALmixer_SetError("No channels are allocated"); | |
4162 retval = -1.0f; | |
4163 } | |
4164 else | |
4165 { | |
4166 retval = running_total / Number_of_Channels_global; | |
4167 } | |
4168 } | |
4169 return retval; | |
4170 } | |
4171 | |
4172 static ALfloat Internal_GetMinVolumeSource(ALuint source) | |
4173 { | |
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4174 ALint channel; |
0 | 4175 if(0 == source) |
4176 { | |
4177 return Internal_GetMinVolumeChannel(-1); | |
4178 } | |
4179 | |
4180 channel = Internal_GetChannel(source); | |
4181 if(-1 == channel) | |
4182 { | |
4183 ALmixer_SetError("Cannot GetVolume: %s", ALmixer_GetError()); | |
4184 return -1.0f; | |
4185 } | |
4186 | |
4187 return Internal_GetMinVolumeChannel(channel); | |
4188 } | |
4189 | |
4190 | |
4191 /* Changes the listener volume */ | |
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4192 static ALboolean Internal_SetMasterVolume(ALfloat volume) |
0 | 4193 { |
4194 ALenum error; | |
4195 alListenerf(AL_GAIN, volume); | |
4196 if((error = alGetError()) != AL_NO_ERROR) | |
4197 { | |
4198 ALmixer_SetError("%s", | |
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4199 alGetString(error) ); |
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4200 return AL_FALSE; |
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4201 } |
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4202 return AL_TRUE; |
0 | 4203 } |
4204 | |
4205 static ALfloat Internal_GetMasterVolume() | |
4206 { | |
4207 ALenum error; | |
4208 ALfloat volume; | |
4209 alGetListenerf(AL_GAIN, &volume); | |
4210 if((error = alGetError()) != AL_NO_ERROR) | |
4211 { | |
4212 ALmixer_SetError("%s", | |
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4213 alGetString(error) ); |
0 | 4214 return -1.0f; |
4215 } | |
4216 return volume; | |
4217 } | |
4218 | |
4219 | |
4220 | |
4221 | |
4222 /* Will fade out currently playing channels. | |
4223 * It starts at the current volume level and goes down */ | |
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4224 static ALint Internal_ExpireChannel(ALint channel, ALint ticks) |
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4225 { |
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4226 ALuint current_time = ALmixer_GetTicks(); |
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4227 ALuint counter = 0; |
0 | 4228 |
4229 /* We can't accept 0 as a value because of div-by-zero. | |
4230 * If zero, just call Halt at normal | |
4231 * volume | |
4232 */ | |
4233 if(0 == ticks) | |
4234 { | |
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4235 return Internal_HaltChannel(channel, AL_TRUE); |
0 | 4236 } |
4237 if(ticks < -1) | |
4238 { | |
4239 ticks = -1; | |
4240 } | |
4241 | |
4242 | |
4243 if(channel >= Number_of_Channels_global) | |
4244 { | |
4245 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); | |
4246 return -1; | |
4247 } | |
4248 | |
4249 if(channel >= 0) | |
4250 { | |
4251 if(ALmixer_Channel_List[channel].channel_in_use) | |
4252 { | |
4253 /* Set expire start time */ | |
4254 ALmixer_Channel_List[channel].start_time = current_time; | |
4255 /* Set the expire ticks */ | |
4256 ALmixer_Channel_List[channel].expire_ticks = ticks; | |
4257 | |
4258 counter++; | |
4259 } | |
4260 } | |
4261 /* Else need to fade out all channels */ | |
4262 else | |
4263 { | |
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4264 ALint i; |
0 | 4265 for(i=0; i<Number_of_Channels_global; i++) |
4266 { | |
4267 if(ALmixer_Channel_List[i].channel_in_use) | |
4268 { | |
4269 /* Set expire start time */ | |
4270 ALmixer_Channel_List[i].start_time = current_time; | |
4271 /* Set the expire ticks */ | |
4272 ALmixer_Channel_List[i].expire_ticks = ticks; | |
4273 | |
4274 counter++; | |
4275 } | |
4276 } /* End for loop */ | |
4277 } | |
4278 return counter; | |
4279 } | |
4280 | |
4281 | |
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4282 static ALint Internal_ExpireSource(ALuint source, ALint ticks) |
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4283 { |
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|
4284 ALint channel; |
0 | 4285 if(0 == source) |
4286 { | |
4287 return Internal_ExpireChannel(-1, ticks); | |
4288 } | |
4289 | |
4290 channel = Internal_GetChannel(source); | |
4291 if(-1 == channel) | |
4292 { | |
4293 ALmixer_SetError("Cannot Expire source: %s", ALmixer_GetError()); | |
4294 return -1; | |
4295 } | |
4296 return Internal_ExpireChannel(channel, ticks); | |
4297 } | |
4298 | |
4299 | |
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4300 static ALint Internal_QueryChannel(ALint channel) |
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4301 { |
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4302 ALint i; |
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4303 ALint counter = 0; |
0 | 4304 if(channel >= Number_of_Channels_global) |
4305 { | |
4306 ALmixer_SetError("Invalid channel: %d", channel); | |
4307 return -1; | |
4308 } | |
4309 | |
4310 if(channel >= 0) | |
4311 { | |
4312 return ALmixer_Channel_List[channel].channel_in_use; | |
4313 } | |
4314 | |
4315 /* Else, return the number of channels in use */ | |
4316 for(i=0; i<Number_of_Channels_global; i++) | |
4317 { | |
4318 if(ALmixer_Channel_List[i].channel_in_use) | |
4319 { | |
4320 counter++; | |
4321 } | |
4322 } | |
4323 return counter; | |
4324 } | |
4325 | |
4326 | |
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4327 static ALint Internal_QuerySource(ALuint source) |
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4328 { |
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4329 ALint channel; |
0 | 4330 if(0 == source) |
4331 { | |
4332 return Internal_QueryChannel(-1); | |
4333 } | |
4334 | |
4335 channel = Internal_GetChannel(source); | |
4336 if(-1 == channel) | |
4337 { | |
4338 ALmixer_SetError("Cannot query source: %s", ALmixer_GetError()); | |
4339 return -1; | |
4340 } | |
4341 | |
4342 return Internal_QueryChannel(channel); | |
4343 } | |
4344 | |
4345 | |
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4346 static ALuint Internal_CountUnreservedUsedChannels() |
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4347 { |
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4348 ALint i; |
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4349 ALuint counter = 0; |
0 | 4350 |
4351 | |
4352 /* Else, return the number of channels in use */ | |
4353 for(i=Number_of_Reserve_Channels_global; i<Number_of_Channels_global; i++) | |
4354 { | |
4355 if(ALmixer_Channel_List[i].channel_in_use) | |
4356 { | |
4357 counter++; | |
4358 } | |
4359 } | |
4360 return counter; | |
4361 } | |
4362 | |
2
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4363 static ALuint Internal_CountUnreservedFreeChannels() |
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4364 { |
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4365 ALint i; |
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4366 ALuint counter = 0; |
0 | 4367 |
4368 | |
4369 /* Else, return the number of channels in use */ | |
4370 for(i=Number_of_Reserve_Channels_global; i<Number_of_Channels_global; i++) | |
4371 { | |
4372 if( ! ALmixer_Channel_List[i].channel_in_use) | |
4373 { | |
4374 counter++; | |
4375 } | |
4376 } | |
4377 return counter; | |
4378 } | |
4379 | |
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4380 static ALuint Internal_CountAllUsedChannels() |
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4381 { |
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4382 ALint i; |
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4383 ALuint counter = 0; |
0 | 4384 |
4385 | |
4386 /* Else, return the number of channels in use */ | |
4387 for(i=0; i<Number_of_Channels_global; i++) | |
4388 { | |
4389 if(ALmixer_Channel_List[i].channel_in_use) | |
4390 { | |
4391 counter++; | |
4392 } | |
4393 } | |
4394 return counter; | |
4395 } | |
4396 | |
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4397 static ALuint Internal_CountAllFreeChannels() |
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4398 { |
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4399 ALint i; |
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4400 ALuint counter = 0; |
0 | 4401 |
4402 | |
4403 /* Else, return the number of channels in use */ | |
4404 for(i=0; i<Number_of_Channels_global; i++) | |
4405 { | |
4406 if( ! ALmixer_Channel_List[i].channel_in_use) | |
4407 { | |
4408 counter++; | |
4409 } | |
4410 } | |
4411 return counter; | |
4412 } | |
4413 | |
4414 | |
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4415 static ALint Internal_PlayingChannel(ALint channel) |
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4416 { |
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4417 ALint i; |
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4418 ALint counter = 0; |
0 | 4419 ALint state; |
4420 | |
4421 if(channel >= Number_of_Channels_global) | |
4422 { | |
4423 ALmixer_SetError("Invalid channel: %d", channel); | |
4424 return -1; | |
4425 } | |
4426 | |
4427 if(channel >= 0) | |
4428 { | |
4429 if(ALmixer_Channel_List[channel].channel_in_use) | |
4430 { | |
4431 alGetSourcei( | |
4432 ALmixer_Channel_List[channel].alsource, | |
4433 AL_SOURCE_STATE, &state | |
4434 ); | |
4435 if(AL_PLAYING == state) | |
4436 { | |
4437 return 1; | |
4438 } | |
4439 } | |
4440 return 0; | |
4441 } | |
4442 | |
4443 /* Else, return the number of channels in use */ | |
4444 for(i=0; i<Number_of_Channels_global; i++) | |
4445 { | |
4446 if(ALmixer_Channel_List[i].channel_in_use) | |
4447 { | |
4448 alGetSourcei( | |
4449 ALmixer_Channel_List[i].alsource, | |
4450 AL_SOURCE_STATE, &state | |
4451 ); | |
4452 if(AL_PLAYING == state) | |
4453 { | |
4454 counter++; | |
4455 } | |
4456 } | |
4457 } | |
4458 return counter; | |
4459 } | |
4460 | |
4461 | |
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4462 static ALint Internal_PlayingSource(ALuint source) |
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4463 { |
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4464 ALint channel; |
0 | 4465 if(0 == source) |
4466 { | |
4467 return Internal_PlayingChannel(-1); | |
4468 } | |
4469 | |
4470 channel = Internal_GetChannel(source); | |
4471 if(-1 == channel) | |
4472 { | |
4473 ALmixer_SetError("Cannot query source: %s", ALmixer_GetError()); | |
4474 return -1; | |
4475 } | |
4476 | |
4477 return Internal_PlayingChannel(channel); | |
4478 } | |
4479 | |
4480 | |
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4481 static ALint Internal_PausedChannel(ALint channel) |
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4482 { |
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4483 ALint i; |
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4484 ALint counter = 0; |
0 | 4485 ALint state; |
4486 | |
4487 if(channel >= Number_of_Channels_global) | |
4488 { | |
4489 ALmixer_SetError("Invalid channel: %d", channel); | |
4490 return -1; | |
4491 } | |
4492 | |
4493 if(channel >= 0) | |
4494 { | |
4495 if(ALmixer_Channel_List[channel].channel_in_use) | |
4496 { | |
4497 alGetSourcei( | |
4498 ALmixer_Channel_List[channel].alsource, | |
4499 AL_SOURCE_STATE, &state | |
4500 ); | |
4501 if(AL_PAUSED == state) | |
4502 { | |
4503 return 1; | |
4504 } | |
4505 } | |
4506 return 0; | |
4507 } | |
4508 | |
4509 /* Else, return the number of channels in use */ | |
4510 for(i=0; i<Number_of_Channels_global; i++) | |
4511 { | |
4512 if(ALmixer_Channel_List[i].channel_in_use) | |
4513 { | |
4514 alGetSourcei( | |
4515 ALmixer_Channel_List[i].alsource, | |
4516 AL_SOURCE_STATE, &state | |
4517 ); | |
4518 if(AL_PAUSED == state) | |
4519 { | |
4520 counter++; | |
4521 } | |
4522 } | |
4523 } | |
4524 return counter; | |
4525 } | |
4526 | |
4527 | |
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4528 static ALint Internal_PausedSource(ALuint source) |
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4529 { |
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4530 ALint channel; |
0 | 4531 if(0 == source) |
4532 { | |
4533 return Internal_PausedChannel(-1); | |
4534 } | |
4535 | |
4536 channel = Internal_GetChannel(source); | |
4537 if(-1 == channel) | |
4538 { | |
4539 ALmixer_SetError("Cannot query source: %s", ALmixer_GetError()); | |
4540 return -1; | |
4541 } | |
4542 | |
4543 return Internal_PausedChannel(channel); | |
4544 } | |
4545 | |
4546 | |
4547 | |
4548 | |
4549 | |
4550 | |
4551 /* Private function for Updating ALmixer. | |
4552 * This is a very big and ugly function. | |
4553 * It should return the number of buffers that were | |
4554 * queued during the call. The value might be | |
4555 * used to guage how long you might wait to | |
4556 * call the next update loop in case you are worried | |
4557 * about preserving CPU cycles. The idea is that | |
4558 * when a buffer is queued, there was probably some | |
4559 * CPU intensive looping which took awhile. | |
4560 * It's mainly provided as a convenience. | |
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4561 * Timing the call with ALmixer_GetTicks() would produce |
0 | 4562 * more accurate information. |
4563 * Returns a negative value if there was an error, | |
4564 * the value being the number of errors. | |
4565 */ | |
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4566 static ALint Update_ALmixer(void* data) |
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4567 { |
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4568 ALint retval = 0; |
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4569 ALint error_flag = 0; |
0 | 4570 ALenum error; |
4571 ALint state; | |
2
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4572 ALint i=0; |
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4573 |
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4574 #ifdef ENABLE_ALMIXER_THREADS |
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4575 SDL_LockMutex(s_simpleLock); |
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4576 #endif |
0 | 4577 if(0 == ALmixer_Initialized) |
4578 { | |
2
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4579 #ifdef ENABLE_ALMIXER_THREADS |
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4580 SDL_UnlockMutex(s_simpleLock); |
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4581 #endif |
0 | 4582 return 0; |
4583 } | |
4584 | |
4585 /* Check the quick flag to see if anything needs updating */ | |
4586 /* If anything is playing, then we have to do work */ | |
4587 if( 0 == Is_Playing_global) | |
4588 { | |
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4589 #ifdef ENABLE_ALMIXER_THREADS |
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4590 SDL_UnlockMutex(s_simpleLock); |
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4591 #endif |
0 | 4592 return 0; |
4593 } | |
4594 /* Clear error */ | |
4595 if((error = alGetError()) != AL_NO_ERROR) | |
4596 { | |
4597 fprintf(stderr, "08Testing errpr before unqueue because getting stuff, for OS X this is expected: %s\n", | |
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4598 alGetString(error)); |
0 | 4599 } |
4600 alGetError(); | |
4601 | |
4602 for(i=0; i<Number_of_Channels_global; i++) | |
4603 { | |
4604 if( ALmixer_Channel_List[i].channel_in_use ) | |
4605 { | |
4606 | |
4607 /* For simplicity, before we do anything else, | |
4608 * we can check the timeout and fading values | |
4609 * and do the appropriate things | |
4610 */ | |
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4611 ALuint current_time = ALmixer_GetTicks(); |
0 | 4612 |
4613 /* Check to see if we need to halt due to Timed play */ | |
4614 if(ALmixer_Channel_List[i].expire_ticks != -1) | |
4615 { | |
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4616 ALuint target_time = (ALuint)ALmixer_Channel_List[i].expire_ticks |
0 | 4617 + ALmixer_Channel_List[i].start_time; |
4618 alGetSourcei(ALmixer_Channel_List[i].alsource, | |
4619 AL_SOURCE_STATE, &state); | |
4620 if((error = alGetError()) != AL_NO_ERROR) | |
4621 { | |
4622 fprintf(stderr, "06Testing errpr before unqueue because getting stuff, for OS X this is expected: %s\n", | |
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4623 alGetString(error)); |
0 | 4624 } |
4625 | |
4626 /* Check the time, and also make sure that it is not | |
4627 * paused (if paused, we don't want to make the | |
4628 * evaluation because when resumed, we will adjust | |
4629 * the times to compensate for the pause). | |
4630 */ | |
4631 if( (current_time >= target_time) | |
4632 && (state != AL_PAUSED) ) | |
4633 { | |
4634 /* Stop the playback */ | |
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4635 Internal_HaltChannel(i, AL_TRUE); |
0 | 4636 if((error = alGetError()) != AL_NO_ERROR) |
4637 { | |
4638 fprintf(stderr, "07Testing errpr before unqueue because getting stuff, for OS X this is expected: %s\n", | |
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4639 alGetString(error)); |
0 | 4640 } |
4641 | |
4642 /* Everything should be done so go on to the next loop */ | |
4643 continue; | |
4644 } | |
4645 } /* End if time expired check */ | |
4646 | |
4647 /* Check to see if we need to adjust the volume for fading */ | |
4648 if( ALmixer_Channel_List[i].fade_enabled ) | |
4649 { | |
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4650 ALuint target_time = ALmixer_Channel_List[i].fade_expire_ticks |
0 | 4651 + ALmixer_Channel_List[i].fade_start_time; |
4652 alGetSourcei(ALmixer_Channel_List[i].alsource, | |
4653 AL_SOURCE_STATE, &state); | |
4654 if((error = alGetError()) != AL_NO_ERROR) | |
4655 { | |
4656 fprintf(stderr, "05Testing errpr before unqueue because getting stuff, for OS X this is expected: %s\n", | |
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4657 alGetString(error)); |
0 | 4658 } |
4659 | |
4660 /* Check the time, and also make sure that it is not | |
4661 * paused (if paused, we don't want to make the | |
4662 * evaluation because when resumed, we will adjust | |
4663 * the times to compensate for the pause). | |
4664 */ | |
4665 if(state != AL_PAUSED) | |
4666 { | |
4667 ALfloat t; | |
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4668 ALuint delta_time; |
0 | 4669 ALfloat current_volume; |
4670 if(current_time >= target_time) | |
4671 { | |
4672 /* Need to constrain value to the end time | |
4673 * (can't go pass the value for calculations) | |
4674 */ | |
4675 current_time = target_time; | |
4676 /* We can disable the fade flag now */ | |
4677 ALmixer_Channel_List[i].fade_enabled = 0; | |
4678 } | |
4679 /* Use the linear interpolation formula: | |
4680 * X = (1-t)x0 + tx1 | |
4681 * where x0 would be the start value | |
4682 * and x1 is the final value | |
4683 * and t is delta_time*inv_time (adjusts 0 <= time <= 1) | |
4684 * delta_time = current_time-start_time | |
4685 * inv_time = 1/ (end_time-start_time) | |
4686 * so t = current_time-start_time / (end_time-start_time) | |
4687 * | |
4688 */ | |
4689 delta_time = current_time - ALmixer_Channel_List[i].fade_start_time; | |
4690 t = (ALfloat) delta_time * ALmixer_Channel_List[i].fade_inv_time; | |
4691 | |
4692 current_volume = (1.0f-t) * ALmixer_Channel_List[i].fade_start_volume | |
4693 + t * ALmixer_Channel_List[i].fade_end_volume; | |
4694 | |
4695 /* Set the volume */ | |
4696 alSourcef(ALmixer_Channel_List[i].alsource, | |
4697 AL_MAX_GAIN, current_volume); | |
4698 if((error = alGetError()) != AL_NO_ERROR) | |
4699 { | |
4700 fprintf(stderr, "04Testing errpr before unqueue because getting stuff, for OS X this is expected: %s\n", | |
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4701 alGetString(error)); |
0 | 4702 } |
4703 | |
4704 /* | |
4705 fprintf(stderr, "Current time =%d\n", current_time); | |
4706 fprintf(stderr, "Current vol=%f on channel %d\n", current_volume, i); | |
4707 */ | |
4708 } /* End if not PAUSED */ | |
4709 } /* End if fade_enabled */ | |
4710 | |
4711 | |
4712 /* Okay, now that the time expired and fading stuff | |
4713 * is done, do the rest of the hard stuff | |
4714 */ | |
4715 | |
4716 | |
4717 /* For predecoded, check to see if done */ | |
4718 if( ALmixer_Channel_List[i].almixer_data->decoded_all ) | |
4719 { | |
4720 | |
4721 #if 0 | |
4722 /********* Remove this **********/ | |
4723 ALint buffers_processed; | |
4724 ALint buffers_still_queued; | |
4725 fprintf(stderr, "For Predecoded\n"); | |
4726 | |
4727 alGetSourcei( | |
4728 ALmixer_Channel_List[i].alsource, | |
4729 AL_SOURCE_STATE, &state | |
4730 ); | |
4731 switch(state) { | |
4732 case AL_PLAYING: | |
4733 fprintf(stderr, "Channel '%d' is PLAYING\n", i); | |
4734 break; | |
4735 case AL_PAUSED: | |
4736 fprintf(stderr, "Channel '%d' is PAUSED\n",i); | |
4737 break; | |
4738 case AL_STOPPED: | |
4739 fprintf(stderr, "Channel '%d' is STOPPED\n",i); | |
4740 break; | |
4741 case AL_INITIAL: | |
4742 fprintf(stderr, "Channel '%d' is INITIAL\n",i); | |
4743 break; | |
4744 default: | |
4745 fprintf(stderr, "Channel '%d' is UNKNOWN\n",i); | |
4746 break; | |
4747 } | |
4748 | |
4749 alGetSourcei( | |
4750 ALmixer_Channel_List[i].alsource, | |
4751 AL_BUFFERS_PROCESSED, &buffers_processed | |
4752 ); | |
4753 fprintf(stderr, "Buffers processed = %d\n", buffers_processed); | |
4754 | |
4755 alGetSourcei( | |
4756 ALmixer_Channel_List[i].alsource, | |
4757 AL_BUFFERS_QUEUED, &buffers_still_queued | |
4758 ); | |
4759 | |
4760 /******** END REMOVE *******/ | |
4761 #endif | |
4762 /* FIXME: Ugh! Somewhere an alError is being thrown ("Invalid Enum Value"), but I can't | |
4763 * find it. It only seems to be thrown for OS X. I placed error messages after every al* | |
4764 * command I could find in the above loops, but the error doesn't seem to show | |
4765 * up until around here. I mistook it for a get queued buffers | |
4766 * error in OS X. I don't think there's an error down there. | |
4767 * For now, I'm clearing the error here. | |
4768 */ | |
4769 | |
4770 if((error = alGetError()) != AL_NO_ERROR) | |
4771 { | |
4772 fprintf(stderr, "03Testing errpr before unqueue because getting stuff, for OS X this is expected: %s\n", | |
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4773 alGetString(error)); |
0 | 4774 } |
4775 | |
4776 | |
4777 alGetSourcei( | |
4778 ALmixer_Channel_List[i].alsource, | |
4779 AL_SOURCE_STATE, &state | |
4780 ); | |
4781 if((error = alGetError()) != AL_NO_ERROR) | |
4782 { | |
4783 fprintf(stderr, "02Testing errpr before unqueue because getting stuff, for OS X this is expected: %s\n", | |
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4784 alGetString(error)); |
0 | 4785 } |
4786 | |
4787 | |
4788 if(AL_STOPPED == state) | |
4789 { | |
4790 /* Playback has ended. | |
4791 * Loop if necessary, or launch callback | |
4792 * and clear channel (or clear channel and | |
4793 * then launch callback?) | |
4794 */ | |
4795 | |
4796 | |
4797 /* Need to check for loops */ | |
4798 if(ALmixer_Channel_List[i].loops != 0) | |
4799 { | |
4800 /* Corner Case: If the buffer has | |
4801 * been modified using Seek, | |
4802 * the loop will start at the seek | |
4803 * position. | |
4804 */ | |
4805 if(ALmixer_Channel_List[i].loops != -1) | |
4806 { | |
4807 ALmixer_Channel_List[i].loops--; | |
4808 } | |
4809 alSourcePlay(ALmixer_Channel_List[i].alsource); | |
4810 if((error = alGetError()) != AL_NO_ERROR) | |
4811 { | |
4812 fprintf(stderr, "50Testing error: %s\n", | |
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4813 alGetString(error)); |
0 | 4814 } |
4815 continue; | |
4816 } | |
4817 /* No loops. End play. */ | |
4818 else | |
4819 { | |
4820 /* Problem: It seems that when mixing | |
4821 * streamed and predecoded sources, | |
4822 * the previous instance lingers, | |
4823 * so we need to force remove | |
4824 * the data from the source. | |
4825 * The sharing problem | |
4826 * occurs when a previous predecoded buffer is played on | |
4827 * a source, and then a streamed source is played later | |
4828 * on that same source. OpenAL isn't consistently | |
4829 * removing the previous buffer so both get played. | |
4830 * (Different dists seem to have different quirks. | |
4831 * The problem might lead to crashes in the worst case.) | |
4832 */ | |
4833 /* Additional problem: There is another | |
4834 * inconsistency among OpenAL distributions. | |
4835 * Both Loki and Creative Windows seem to keep | |
4836 * the buffer queued which requires removing. | |
4837 * But the Creative Macintosh version does | |
4838 * not have any buffer queued after play | |
4839 * and it returns the error: Invalid Enum Value | |
4840 * if I try to unqueue it. | |
4841 * So I'm going to put in a check to see if I | |
4842 * can detect any buffers queued first | |
4843 * and then unqueue them if I can see them. | |
4844 * Additional note: The new CoreAudio based | |
4845 * implementation leaves it's buffer queued | |
4846 * like Loki and Creative Windows. But | |
4847 * considering all the problems I'm having | |
4848 * with the different distributions, this | |
4849 * check seems reasonable. | |
4850 */ | |
4851 ALint buffers_still_queued; | |
4852 if((error = alGetError()) != AL_NO_ERROR) | |
4853 { | |
4854 fprintf(stderr, "01Testing errpr before unqueue because getting stuff, for OS X this is expected: %s\n", | |
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4855 alGetString(error)); |
0 | 4856 } |
4857 | |
4858 alGetSourcei( | |
4859 ALmixer_Channel_List[i].alsource, | |
4860 AL_BUFFERS_QUEUED, &buffers_still_queued | |
4861 ); | |
4862 if((error = alGetError()) != AL_NO_ERROR) | |
4863 { | |
4864 fprintf(stderr, "Error with unqueue, for OS X this is expected: %s\n", | |
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4865 alGetString(error)); |
0 | 4866 ALmixer_SetError("Failed detecting unqueued predecoded buffer (expected with OS X): %s", |
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4867 alGetString(error) ); |
0 | 4868 error_flag--; |
4869 } | |
4870 if(buffers_still_queued > 0) | |
4871 { | |
1 | 4872 |
4873 #if 0 /* This triggers an error in OS X Core Audio. */ | |
4874 alSourceUnqueueBuffers( | |
4875 ALmixer_Channel_List[i].alsource, | |
4876 1, | |
4877 ALmixer_Channel_List[i].almixer_data->buffer | |
4878 ); | |
4879 #else | |
4880 /* fprintf(stderr, "In the Bob Aron section...about to clear source\n"); | |
0 | 4881 PrintQueueStatus(ALmixer_Channel_List[i].alsource); |
1 | 4882 */ |
0 | 4883 /* Rather than force unqueuing the buffer, let's see if |
4884 * setting the buffer to none works (the OpenAL 1.0 | |
4885 * Reference Annotation suggests this should work). | |
4886 */ | |
4887 alSourcei(ALmixer_Channel_List[i].alsource, | |
4888 AL_BUFFER, AL_NONE); | |
4889 /* | |
4890 PrintQueueStatus(ALmixer_Channel_List[i].alsource); | |
4891 */ | |
1 | 4892 #endif |
0 | 4893 if((error = alGetError()) != AL_NO_ERROR) |
4894 { | |
4895 fprintf(stderr, "Error with unqueue, after alSourceUnqueueBuffers, buffers_still_queued=%d, error is: %s", buffers_still_queued, | |
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4896 alGetString(error)); |
0 | 4897 ALmixer_SetError("Predecoded Unqueue buffer failed: %s", |
2
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4898 alGetString(error) ); |
0 | 4899 error_flag--; |
4900 } | |
4901 | |
4902 } | |
4903 | |
4904 Clean_Channel(i); | |
4905 /* Subtract counter */ | |
4906 Is_Playing_global--; | |
4907 | |
4908 /* Launch callback */ | |
2
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4909 Invoke_Channel_Done_Callback(i, AL_TRUE); |
0 | 4910 |
4911 /* We're done for this loop. | |
4912 * Go to next channel | |
4913 */ | |
4914 continue; | |
4915 } | |
4916 continue; | |
4917 } | |
4918 } /* End if decoded_all */ | |
4919 /* For streamed */ | |
4920 else | |
4921 { | |
4922 ALint buffers_processed; | |
4923 ALint buffers_still_queued; | |
4924 ALint current_buffer_id; | |
4925 | |
4926 ALuint unqueued_buffer_id; | |
4927 #if 0 | |
4928 /********* Remove this **********/ | |
4929 fprintf(stderr, "For Streamed\n"); | |
4930 | |
4931 alGetSourcei( | |
4932 ALmixer_Channel_List[i].alsource, | |
4933 AL_SOURCE_STATE, &state | |
4934 ); | |
4935 switch(state) { | |
4936 case AL_PLAYING: | |
4937 fprintf(stderr, "Channel '%d' is PLAYING\n", i); | |
4938 break; | |
4939 case AL_PAUSED: | |
4940 fprintf(stderr, "Channel '%d' is PAUSED\n",i); | |
4941 break; | |
4942 case AL_STOPPED: | |
4943 fprintf(stderr, "Channel '%d' is STOPPED\n",i); | |
4944 break; | |
4945 case AL_INITIAL: | |
4946 fprintf(stderr, "Channel '%d' is INITIAL\n",i); | |
4947 break; | |
4948 default: | |
4949 fprintf(stderr, "Channel '%d' is UNKNOWN\n",i); | |
4950 break; | |
4951 } | |
4952 /******** END REMOVE *******/ | |
4953 #endif | |
4954 /* Get the number of buffers still queued */ | |
4955 alGetSourcei( | |
4956 ALmixer_Channel_List[i].alsource, | |
4957 AL_BUFFERS_QUEUED, &buffers_still_queued | |
4958 ); | |
4959 if((error = alGetError()) != AL_NO_ERROR) | |
4960 { | |
4961 fprintf(stderr, "51Testing error: %s\n", | |
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4962 alGetString(error)); |
0 | 4963 } |
4964 /* Get the number of buffers processed | |
4965 * so we know if we need to refill | |
4966 */ | |
2
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4967 /* WARNING: It looks like Snow Leopard some times crashes on this call under x86_64 |
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4968 * typically when I suffer a lot of buffer underruns. |
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4969 */ |
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4970 // fprintf(stderr, "calling AL_BUFFERS_PROCESSED on source:%d", ALmixer_Channel_List[i].alsource); |
0 | 4971 alGetSourcei( |
4972 ALmixer_Channel_List[i].alsource, | |
4973 AL_BUFFERS_PROCESSED, &buffers_processed | |
4974 ); | |
4975 if((error = alGetError()) != AL_NO_ERROR) | |
4976 { | |
4977 fprintf(stderr, "52Testing error: %s\n", | |
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4978 alGetString(error)); |
0 | 4979 } |
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4980 // fprintf(stderr, "finished AL_BUFFERS_PROCESSED, buffers_processed=%d", buffers_processed); |
0 | 4981 |
4982 /* WTF!!! The Nvidia distribution is failing on the alGetSourcei(source, AL_BUFFER, buf_id) call. | |
4983 * I need this call to figure out which buffer OpenAL is currently playing. | |
4984 * It keeps returning an "Invalid Enum" error. | |
4985 * This is totally inane! It's a basic query. | |
4986 * By the spec, this functionality is not explicitly defined so Nvidia refuses to | |
4987 * fix this behavior, even though all other distributions work fine with this. | |
4988 * The only workaround for this is for | |
4989 * a significant rewrite of my code which requires me to | |
4990 * duplicate the OpenAL queued buffers state with my own | |
4991 * code and try to derive what the current playing buffer is by indirect observation of | |
4992 * looking at buffers_processed. But of course this has a ton of downsides since my | |
4993 * queries do not give me perfect timing of what OpenAL is actually doing and | |
4994 * the fact that some of the distributions seem to have buffer queuing problems | |
4995 * with their query results (CoreAudio). This also means a ton of extra code | |
4996 * on my side. The lack of support of a 1 line call has required me to | |
4997 * implement yet another entire state machine. <sigh> | |
4998 */ | |
4999 #if 0 /* This code will not work until possibly OpenAL 1.1 because of Nvidia */ | |
5000 /* Get the id to the current buffer playing */ | |
5001 alGetSourcei( | |
5002 ALmixer_Channel_List[i].alsource, | |
5003 AL_BUFFER, ¤t_buffer_id | |
5004 ); | |
5005 if((error = alGetError()) != AL_NO_ERROR) | |
5006 { | |
5007 fprintf(stderr, "53Testing error: %s\n", | |
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5008 alGetString(error)); |
0 | 5009 } |
5010 | |
5011 /* Before the hard stuff, check to see if the | |
5012 * current queued AL buffer has changed. | |
5013 * If it has, we should launch a data callback if | |
5014 * necessary | |
5015 */ | |
5016 if( ((ALuint)current_buffer_id) != | |
5017 ALmixer_Channel_List[i].almixer_data->current_buffer) | |
5018 { | |
5019 ALmixer_Channel_List[i].almixer_data->current_buffer | |
5020 = (ALuint)current_buffer_id; | |
5021 | |
5022 Invoke_Streamed_Channel_Data_Callback(i, ALmixer_Channel_List[i].almixer_data, current_buffer_id); | |
5023 } | |
5024 #else | |
5025 /* Only do this if "access_data" was requested (i.e. the circular_buffer!=NULL) | |
5026 * And if one of the two are true: | |
5027 * Either buffers_processed > 0 (because the current_buffer might have changed) | |
5028 * or if the current_buffer==0 (because we are in an initial state or recovering from | |
5029 * a buffer underrun) | |
5030 */ | |
5031 if((ALmixer_Channel_List[i].almixer_data->circular_buffer_queue != NULL) | |
5032 && ( | |
5033 (buffers_processed > 0) || (0 == ALmixer_Channel_List[i].almixer_data->current_buffer) | |
5034 ) | |
5035 ) | |
5036 { | |
2
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5037 ALint k; |
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5038 ALuint queue_ret_flag; |
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5039 ALubyte is_out_of_sync = 0; |
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5040 ALuint my_queue_size = CircularQueueUnsignedInt_Size(ALmixer_Channel_List[i].almixer_data->circular_buffer_queue); |
1 | 5041 /* Ugh, I have to deal with signed/unsigned mismatch here. */ |
5042 ALint buffers_unplayed_int = buffers_still_queued - buffers_processed; | |
2
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5043 ALuint unplayed_buffers; |
1 | 5044 if(buffers_unplayed_int < 0) |
5045 { | |
5046 unplayed_buffers = 0; | |
5047 } | |
5048 else | |
5049 { | |
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5050 unplayed_buffers = (ALuint)buffers_unplayed_int; |
1 | 5051 } |
0 | 5052 /* |
5053 fprintf(stderr, "Queue in processed check, before pop, buffers_processed=%d\n", buffers_processed); | |
5054 CircularQueueUnsignedInt_Print(ALmixer_Channel_List[i].almixer_data->circular_buffer_queue); | |
5055 */ | |
5056 /* We can't make any determinations solely based on the number of buffers_processed | |
5057 * because currently, we only unqueue 1 buffer per loop. That means if 2 or more | |
5058 * buffers became processed in one loop, the following loop, we would have | |
5059 * at least that_many-1 buffers_processed (plus possible new processed). | |
5060 * If we tried to just remove 1 buffer from our queue, we would be incorrect | |
5061 * because we would not actually reflect the current playing buffer. | |
5062 * So the solution seems to be to make sure our queue is the same size | |
5063 * as the number of buffers_queued-buffers_processed, and return the head of our queue | |
5064 * as the current playing buffer. | |
5065 */ | |
5066 /* Also, we have a corner case. When we first start playing or if we have | |
5067 * a buffer underrun, we have not done a data callback. | |
5068 * In this case, we need to see if there is any new data in our queue | |
5069 * and if so, launch that data callback. | |
5070 */ | |
5071 /* Warning, this code risks the possibility of no data callback being fired if | |
5072 * the system is really late (or skipped buffers). | |
5073 */ | |
5074 | |
5075 /* First, let's syncronize our queue with the OpenAL queue */ | |
5076 #if 0 | |
5077 fprintf(stderr, "inside, Buffers processed=%d, Buffers queued=%d, my queue=%d\n", | |
5078 buffers_processed, buffers_still_queued, my_queue_size); | |
1 | 5079 #endif |
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5080 is_out_of_sync = 1; |
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5081 for(k=0; k<buffers_processed; k++) |
0 | 5082 { |
2
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5083 queue_ret_flag = CircularQueueUnsignedInt_PopFront( |
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5084 ALmixer_Channel_List[i].almixer_data->circular_buffer_queue); |
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5085 if(0 == queue_ret_flag) |
0 | 5086 { |
2
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5087 fprintf(stderr, "53 Error popping queue\n"); |
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5088 } |
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5089 } |
0 | 5090 my_queue_size = CircularQueueUnsignedInt_Size(ALmixer_Channel_List[i].almixer_data->circular_buffer_queue); |
5091 /* We have several possibilities we need to handle: | |
5092 * 1) We are in an initial state or underrun and need to do a data callback on the head. | |
5093 * 2) We were out of sync and need to do a new data callback on the new head. | |
5094 * 3) We were not out of sync but just had left over processed buffers which caused us to | |
5095 * fall in this block of code. (Don't do anything.) | |
5096 */ | |
5097 if( (0 == ALmixer_Channel_List[i].almixer_data->current_buffer) || (1 == is_out_of_sync) ) | |
5098 { | |
5099 if(my_queue_size > 0) | |
5100 { | |
5101 current_buffer_id = CircularQueueUnsignedInt_Front( | |
5102 ALmixer_Channel_List[i].almixer_data->circular_buffer_queue); | |
5103 if(0 == current_buffer_id) | |
5104 { | |
5105 fprintf(stderr, "53a Internal Error, current_buffer_id=0 when it shouldn't be 0\n"); | |
5106 } | |
5107 /* | |
5108 else | |
5109 { | |
5110 fprintf(stderr, "Queue in processed check, after pop\n"); | |
5111 CircularQueueUnsignedInt_Print(ALmixer_Channel_List[i].almixer_data->circular_buffer_queue); | |
5112 } | |
5113 */ | |
5114 ALmixer_Channel_List[i].almixer_data->current_buffer | |
5115 = (ALuint)current_buffer_id; | |
5116 | |
5117 #if 0 | |
5118 /* Remove me...only for checking...doesn't work on Nvidia */ | |
5119 { | |
5120 ALuint real_id; | |
5121 alGetSourcei( | |
5122 ALmixer_Channel_List[i].alsource, | |
5123 AL_BUFFER, &real_id | |
5124 ); | |
5125 alGetError(); | |
5126 fprintf(stderr, "Callback fired on data buffer=%d, real_id shoud be=%d\n", current_buffer_id, real_id); | |
5127 } | |
5128 #endif | |
5129 Invoke_Streamed_Channel_Data_Callback(i, ALmixer_Channel_List[i].almixer_data, current_buffer_id); | |
5130 } | |
5131 else | |
5132 { | |
1 | 5133 /* |
0 | 5134 fprintf(stderr, "53b, Notice/Warning:, OpenAL queue has been depleted.\n"); |
1 | 5135 PrintQueueStatus(ALmixer_Channel_List[i].alsource); |
5136 */ | |
0 | 5137 /* In this case, we might either be in an underrun or finished with playback */ |
5138 ALmixer_Channel_List[i].almixer_data->current_buffer = 0; | |
5139 } | |
5140 } | |
5141 } | |
5142 #endif | |
5143 | |
5144 | |
5145 | |
5146 /* Just a test - remove | |
5147 if( ALmixer_Channel_List[i].loops > 0) | |
5148 { | |
5149 fprintf(stderr, ">>>>>>>>>>>>>>>Loops = %d\n", | |
5150 ALmixer_Channel_List[i].loops); | |
5151 } | |
5152 */ | |
5153 #if 0 | |
5154 fprintf(stderr, "Buffers processed = %d\n", buffers_processed); | |
5155 fprintf(stderr, "Buffers queued= %d\n", buffers_still_queued); | |
5156 #endif | |
5157 /* We've used up a buffer so we need to unqueue and replace */ | |
5158 /* Okay, it gets more complicated here: | |
5159 * We need to Queue more data | |
5160 * if buffers_processed > 0 or | |
5161 * if num_of_buffers_in_use < NUMBER_OF_QUEUE_BUFFERS | |
5162 * but we don't do this if at EOF, | |
5163 * except when there is looping | |
5164 */ | |
5165 /* For this to work, we must rely on EVERYTHING | |
5166 * else to unset the EOF if there is looping. | |
5167 * Remember, even Play() must do this | |
5168 */ | |
5169 | |
5170 /* If not EOF, then we are still playing. | |
5171 * Inside, we might find num_of_buffers < NUM...QUEUE_BUF.. | |
5172 * or buffers_process > 0 | |
5173 * in which case we queue up. | |
5174 * We also might find no buffers we need to fill, | |
5175 * in which case we just keep going | |
5176 */ | |
5177 if( ! ALmixer_Channel_List[i].almixer_data->eof) | |
5178 { | |
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5179 ALuint bytes_returned; |
0 | 5180 /* We have a priority. We first must assign |
5181 * unused buffers in reserve. If there is nothing | |
5182 * left, then we may unqueue buffers. We can't | |
5183 * do it the other way around because we will | |
5184 * lose the pointer to the unqueued buffer | |
5185 */ | |
5186 if(ALmixer_Channel_List[i].almixer_data->num_buffers_in_use | |
5187 < ALmixer_Channel_List[i].almixer_data->max_queue_buffers) | |
5188 { | |
1 | 5189 #if 0 |
0 | 5190 fprintf(stderr, "Getting more data in NOT_EOF and num_buffers_in_use (%d) < max_queue (%d)\n", |
5191 ALmixer_Channel_List[i].almixer_data->num_buffers_in_use, | |
5192 ALmixer_Channel_List[i].almixer_data->max_queue_buffers); | |
1 | 5193 #endif |
0 | 5194 /* Going to add an unused packet. |
5195 * Grab next packet */ | |
5196 bytes_returned = GetMoreData( | |
5197 ALmixer_Channel_List[i].almixer_data, | |
5198 ALmixer_Channel_List[i].almixer_data->buffer[ | |
5199 ALmixer_Channel_List[i].almixer_data->num_buffers_in_use] | |
5200 ); | |
5201 } | |
5202 /* For processed > 0 */ | |
5203 else if(buffers_processed > 0) | |
5204 { | |
5205 /* Unqueue only 1 buffer for now. | |
5206 * If there are more than one, | |
5207 * let the next Update pass deal with it | |
5208 * so we don't stall the program for too long. | |
5209 */ | |
5210 #if 0 | |
5211 fprintf(stderr, "About to Unqueue, Buffers processed = %d\n", buffers_processed); | |
5212 fprintf(stderr, "Buffers queued= %d\n", buffers_still_queued); | |
5213 fprintf(stderr, "Unqueuing a buffer\n"); | |
5214 #endif | |
5215 alSourceUnqueueBuffers( | |
5216 ALmixer_Channel_List[i].alsource, | |
5217 1, &unqueued_buffer_id | |
5218 ); | |
5219 if((error = alGetError()) != AL_NO_ERROR) | |
5220 { | |
5221 fprintf(stderr, "Error with unqueue: %s", | |
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5222 alGetString(error)); |
0 | 5223 ALmixer_SetError("Unqueue buffer failed: %s", |
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5224 alGetString(error) ); |
0 | 5225 error_flag--; |
5226 } | |
5227 /* | |
5228 fprintf(stderr, "Right after unqueue..."); | |
5229 PrintQueueStatus(ALmixer_Channel_List[i].alsource); | |
5230 fprintf(stderr, "Getting more data for NOT_EOF, max_buffers filled\n"); | |
5231 */ | |
5232 /* Grab unqueued packet */ | |
5233 bytes_returned = GetMoreData( | |
5234 ALmixer_Channel_List[i].almixer_data, | |
5235 unqueued_buffer_id); | |
5236 } | |
5237 /* We are still streaming, but currently | |
5238 * don't need to fill any buffers */ | |
5239 else | |
5240 { | |
5241 /* Might want to check state */ | |
5242 /* In case the playback stopped, | |
2
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5243 * we need to resume |
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5244 * a.k.a. buffer underrun |
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5245 */ |
1 | 5246 #if 1 |
5247 /* Try not refetching the state here because I'm getting a duplicate | |
5248 buffer playback (hiccup) */ | |
0 | 5249 alGetSourcei( |
5250 ALmixer_Channel_List[i].alsource, | |
5251 AL_SOURCE_STATE, &state | |
5252 ); | |
1 | 5253 if((error = alGetError()) != AL_NO_ERROR) |
5254 { | |
5255 fprintf(stderr, "54bTesting error: %s\n", | |
2
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5256 alGetString(error)); |
1 | 5257 } |
5258 /* Get the number of buffers processed | |
5259 */ | |
5260 alGetSourcei( | |
5261 ALmixer_Channel_List[i].alsource, | |
5262 AL_BUFFERS_PROCESSED, | |
5263 &buffers_processed | |
5264 ); | |
5265 if((error = alGetError()) != AL_NO_ERROR) | |
5266 { | |
5267 fprintf(stderr, "54cError, Can't get buffers_processed: %s\n", | |
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5268 alGetString(error)); |
1 | 5269 } |
5270 #endif | |
0 | 5271 if(AL_STOPPED == state) |
5272 { | |
5273 /* Resuming in not eof, but nothing to buffer */ | |
1 | 5274 |
5275 /* Okay, here's another lately discovered problem: | |
5276 * I can't find it in the spec, but for at least some of the | |
5277 * implementations, if I call play on a stopped source that | |
5278 * has processed buffers, all those buffers get marked as unprocessed | |
5279 * on alSourcePlay. So if I had a queue of 25 with 24 of the buffers | |
5280 * processed, on resume, the earlier 24 buffers will get replayed, | |
5281 * causing a "hiccup" like sound in the playback. | |
5282 * To avoid this, I must unqueue all processed buffers before | |
5283 * calling play. But to complicate things, I need to worry about resyncing | |
5284 * the circular queue with this since I designed this thing | |
5285 * with some correlation between the two. However, I might | |
5286 * have already handled this, so I will try writing this code without | |
5287 * syncing for now. | |
5288 * There is currently an assumption that a buffer | |
5289 * was queued above so I actually have something | |
5290 * to play. | |
5291 */ | |
5292 ALint temp_count; | |
2
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5293 #if 0 |
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5294 fprintf(stderr, "STOPPED1, need to clear processed=%d, status is:\n", buffers_processed); |
1 | 5295 PrintQueueStatus(ALmixer_Channel_List[i].alsource); |
2
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5296 #endif |
1 | 5297 for(temp_count=0; temp_count<buffers_processed; temp_count++) |
5298 { | |
5299 alSourceUnqueueBuffers( | |
5300 ALmixer_Channel_List[i].alsource, | |
5301 1, &unqueued_buffer_id | |
5302 ); | |
5303 if((error = alGetError()) != AL_NO_ERROR) | |
5304 { | |
5305 fprintf(stderr, "55aTesting error: %s\n", | |
2
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5306 alGetString(error)); |
1 | 5307 error_flag--; |
5308 } | |
5309 } | |
2
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5310 #if 0 |
1 | 5311 fprintf(stderr, "After unqueue clear...:\n"); |
5312 PrintQueueStatus(ALmixer_Channel_List[i].alsource); | |
2
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5313 #endif |
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5314 /* My assertion: We are STOPPED but not EOF. |
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5315 * This means we have a buffer underrun. |
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5316 * We just cleared out the unqueued buffers. |
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5317 * So we need to reset the mixer_data to reflect we have |
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5318 * no buffers in queue. |
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5319 * We need to GetMoreData and then queue up the data. |
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5320 * Then we need to resume playing. |
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5321 */ |
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5322 #if 0 |
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5323 int buffers_queued; |
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5324 alGetSourcei( |
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5325 ALmixer_Channel_List[i].alsource, |
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5326 AL_BUFFERS_QUEUED, |
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5327 &buffers_queued |
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5328 ); |
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5329 |
1 | 5330 if((error = alGetError()) != AL_NO_ERROR) |
5331 { | |
2
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5332 fprintf(stderr, "Error in PrintQueueStatus, Can't get buffers_queued: %s\n", |
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5333 alGetString(error)); |
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5334 } |
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5335 assert(buffers_queued == 0); |
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5336 fprintf(stderr, "buffer underrun: buffers_queued:%d\n", buffers_queued); |
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5337 #endif |
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5338 |
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5339 /* Reset the number of buffers in use to 0 */ |
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5340 ALmixer_Channel_List[i].almixer_data->num_buffers_in_use = 0; |
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5341 |
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5342 /* Get more data and put it in the first buffer */ |
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5343 bytes_returned = GetMoreData( |
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5344 ALmixer_Channel_List[i].almixer_data, |
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5345 ALmixer_Channel_List[i].almixer_data->buffer[0] |
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5346 ); |
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5347 /* NOTE: We might want to look for EOF and handle it here. |
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5348 * Currently, I just let the next loop handle it which seems to be working. |
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5349 */ |
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5350 if(bytes_returned > 0) |
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5351 { |
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5352 /* Queue up the new data */ |
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5353 alSourceQueueBuffers( |
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5354 ALmixer_Channel_List[i].alsource, |
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5355 1, |
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5356 &ALmixer_Channel_List[i].almixer_data->buffer[0] |
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5357 ); |
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5358 if((error = alGetError()) != AL_NO_ERROR) |
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5359 { |
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5360 fprintf(stderr, "56e alSourceQueueBuffers error: %s\n", |
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5361 alGetString(error)); |
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5362 } |
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5363 /* Increment the number of buffers in use */ |
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5364 ALmixer_Channel_List[i].almixer_data->num_buffers_in_use++; |
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5365 |
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5366 |
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5367 /* We need to empty and update the circular buffer queue if it is in use */ |
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5368 if(ALmixer_Channel_List[i].almixer_data->circular_buffer_queue != NULL) |
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5369 { |
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5370 ALuint queue_ret_flag; |
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5371 CircularQueueUnsignedInt_Clear(ALmixer_Channel_List[i].almixer_data->circular_buffer_queue); |
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5372 queue_ret_flag = CircularQueueUnsignedInt_PushBack( |
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5373 ALmixer_Channel_List[i].almixer_data->circular_buffer_queue, |
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5374 ALmixer_Channel_List[i].almixer_data->buffer[0] |
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5375 ); |
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5376 if(0 == queue_ret_flag) |
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5377 { |
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5378 fprintf(stderr, "56fSerious internal error: CircularQueue could not push into queue.\n"); |
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5379 ALmixer_SetError("Serious internal error: CircularQueue failed to push into queue"); |
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5380 } |
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5381 } |
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5382 |
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5383 |
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5384 |
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5385 |
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5386 /* Resume playback from underrun */ |
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|
5387 alSourcePlay(ALmixer_Channel_List[i].alsource); |
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|
5388 if((error = alGetError()) != AL_NO_ERROR) |
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|
5389 { |
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|
5390 fprintf(stderr, "55Tbesting error: %s\n", |
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|
5391 alGetString(error)); |
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|
5392 } |
1 | 5393 } |
2
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5394 |
0 | 5395 } |
5396 /* Let's escape to the next loop. | |
5397 * All code below this point is for queuing up | |
5398 */ | |
5399 /* | |
1 | 5400 fprintf(stderr, "Entry: Nothing to do...continue\n\n"); |
5401 */ | |
0 | 5402 continue; |
5403 } | |
5404 /* We now know we have to fill an available | |
5405 * buffer. | |
5406 */ | |
5407 | |
5408 /* In the previous branch, we just grabbed more data. | |
5409 * Let's check it to make sure it's okay, | |
5410 * and then queue it up | |
5411 */ | |
2
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5412 /* This check doesn't work anymore because it is now ALuint */ |
0 | 5413 #if 0 |
5414 if(-1 == bytes_returned) | |
5415 { | |
5416 /* Problem occurred...not sure what to do */ | |
5417 /* Go to next loop? */ | |
5418 error_flag--; | |
5419 /* Set the eof flag to force a quit so | |
5420 * we don't get stuck in an infinite loop | |
5421 */ | |
5422 ALmixer_Channel_List[i].almixer_data->eof = 1; | |
5423 continue; | |
5424 } | |
5425 #endif | |
5426 /* This is a special case where we've run | |
5427 * out of data. We should check for loops | |
5428 * and get more data. If there is no loop, | |
5429 * then do nothing and wait for future | |
5430 * update passes to handle the EOF. | |
5431 * The advantage of handling the loop here | |
5432 * instead of waiting for play to stop is | |
5433 * that we should be able to keep the buffer | |
5434 * filled. | |
5435 */ | |
5436 #if 0 | |
5437 else if(0 == bytes_returned) | |
5438 #endif | |
5439 if(0 == bytes_returned) | |
5440 { | |
5441 fprintf(stderr, "We got 0 bytes from reading. Checking for loops\n"); | |
5442 /* Check for loops */ | |
5443 if( ALmixer_Channel_List[i].loops != 0 ) | |
5444 { | |
5445 /* We have to loop, so rewind | |
5446 * and fetch more data | |
5447 */ | |
5448 fprintf(stderr, "Rewinding data\n"); | |
5449 if(0 == Sound_Rewind( | |
5450 ALmixer_Channel_List[i].almixer_data->sample)) | |
5451 { | |
5452 fprintf(stderr, "Rewinding failed\n"); | |
5453 ALmixer_SetError( Sound_GetError() ); | |
5454 ALmixer_Channel_List[i].loops = 0; | |
5455 error_flag--; | |
5456 /* We'll continue on because we do have some valid data */ | |
5457 continue; | |
5458 } | |
5459 /* Remember to reset the data->eof flag */ | |
5460 ALmixer_Channel_List[i].almixer_data->eof = 0; | |
5461 if(ALmixer_Channel_List[i].loops > 0) | |
5462 { | |
5463 ALmixer_Channel_List[i].loops--; | |
5464 } | |
5465 /* Try grabbing another packet now. | |
5466 * Since we may have already unqueued a | |
5467 * buffer, we don't want to lose it. | |
5468 */ | |
5469 if(ALmixer_Channel_List[i].almixer_data->num_buffers_in_use | |
5470 < ALmixer_Channel_List[i].almixer_data->max_queue_buffers) | |
5471 { | |
5472 fprintf(stderr, "We got %d bytes from reading loop. Filling unused packet\n", bytes_returned); | |
5473 /* Grab next packet */ | |
5474 bytes_returned = GetMoreData( | |
5475 ALmixer_Channel_List[i].almixer_data, | |
5476 ALmixer_Channel_List[i].almixer_data->buffer[ | |
5477 ALmixer_Channel_List[i].almixer_data->num_buffers_in_use] | |
5478 ); | |
5479 fprintf(stderr, "We reread %d bytes into unused packet\n", bytes_returned); | |
5480 } | |
5481 /* Refilling unqueued packet */ | |
5482 else | |
5483 { | |
5484 fprintf(stderr, "We got %d bytes from reading loop. Filling unqueued packet\n", bytes_returned); | |
5485 /* Grab next packet */ | |
5486 bytes_returned = GetMoreData( | |
5487 ALmixer_Channel_List[i].almixer_data, | |
5488 unqueued_buffer_id); | |
5489 fprintf(stderr, "We reread %d bytes into unqueued packet\n", bytes_returned); | |
5490 } | |
5491 /* Another error check */ | |
5492 /* | |
5493 if(bytes_returned <= 0) | |
5494 */ | |
5495 if(0 == bytes_returned) | |
5496 { | |
5497 fprintf(stderr, "??????????ERROR\n"); | |
5498 ALmixer_SetError("Could not loop because after rewind, no data could be retrieved"); | |
5499 /* Problem occurred...not sure what to do */ | |
5500 /* Go to next loop? */ | |
5501 error_flag--; | |
5502 /* Set the eof flag to force a quit so | |
5503 * we don't get stuck in an infinite loop | |
5504 */ | |
5505 ALmixer_Channel_List[i].almixer_data->eof = 1; | |
5506 continue; | |
5507 } | |
5508 /* We made it to the end. We still need | |
5509 * to BufferData, so let this branch | |
5510 * fall into the next piece of | |
5511 * code below which will handle that | |
5512 */ | |
5513 | |
5514 | |
5515 } /* END loop check */ | |
5516 else | |
5517 { | |
5518 /* No more loops to do. | |
5519 * EOF flag should be set. | |
5520 * Just go to next loop and | |
5521 * let things be handled correctly | |
5522 * in future update calls | |
5523 */ | |
1 | 5524 /* |
0 | 5525 fprintf(stderr, "SHOULD BE EOF\n"); |
5526 | |
5527 PrintQueueStatus(ALmixer_Channel_List[i].alsource); | |
1 | 5528 */ |
0 | 5529 continue; |
5530 } | |
5531 } /* END if bytes_returned == 0 */ | |
5532 /********* Possible trouble point. I might be queueing empty buffers on the mac. | |
5533 * This check doesn't say if the buffer is valid. Only the EOF assumption is a clue at this point | |
5534 */ | |
5535 /* Fall here */ | |
5536 /* Everything is normal. We aren't | |
5537 * at an EOF, but need to simply | |
5538 * queue more data. The data is already checked for good, | |
5539 * so queue it up */ | |
5540 if(ALmixer_Channel_List[i].almixer_data->num_buffers_in_use | |
5541 < ALmixer_Channel_List[i].almixer_data->max_queue_buffers) | |
5542 { | |
5543 /* Keep count of how many buffers we have | |
5544 * to queue so we can return the value | |
5545 */ | |
5546 retval++; | |
1 | 5547 /* |
0 | 5548 fprintf(stderr, "NOT_EOF???, about to Queue more data for num_buffers (%d) < max_queue (%d)\n", |
5549 ALmixer_Channel_List[i].almixer_data->num_buffers_in_use, | |
5550 ALmixer_Channel_List[i].almixer_data->max_queue_buffers); | |
1 | 5551 */ |
0 | 5552 alSourceQueueBuffers( |
5553 ALmixer_Channel_List[i].alsource, | |
5554 1, | |
5555 &ALmixer_Channel_List[i].almixer_data->buffer[ | |
5556 ALmixer_Channel_List[i].almixer_data->num_buffers_in_use] | |
5557 ); | |
5558 if((error = alGetError()) != AL_NO_ERROR) | |
5559 { | |
5560 fprintf(stderr, "56Testing error: %s\n", | |
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5561 alGetString(error)); |
0 | 5562 } |
5563 /* This is part of the hideous Nvidia workaround. In order to figure out | |
5564 * which buffer to show during callbacks (for things like | |
5565 * o-scopes), I must keep a copy of the buffers that are queued in my own | |
5566 * data structure. This code will be called only if | |
5567 * "access_data" was set, indicated by whether the queue is NULL. | |
5568 */ | |
5569 if(ALmixer_Channel_List[i].almixer_data->circular_buffer_queue != NULL) | |
5570 { | |
2
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|
5571 ALuint queue_ret_flag; |
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|
5572 // fprintf(stderr, "56d: CircularQueue_PushBack.\n"); |
0 | 5573 queue_ret_flag = CircularQueueUnsignedInt_PushBack( |
5574 ALmixer_Channel_List[i].almixer_data->circular_buffer_queue, | |
5575 ALmixer_Channel_List[i].almixer_data->buffer[ALmixer_Channel_List[i].almixer_data->num_buffers_in_use] | |
5576 ); | |
5577 if(0 == queue_ret_flag) | |
5578 { | |
5579 fprintf(stderr, "56aSerious internal error: CircularQueue could not push into queue.\n"); | |
5580 ALmixer_SetError("Serious internal error: CircularQueue failed to push into queue"); | |
5581 } | |
5582 /* | |
5583 else | |
5584 { | |
5585 CircularQueueUnsignedInt_Print(ALmixer_Channel_List[i].almixer_data->circular_buffer_queue); | |
5586 } | |
5587 */ | |
5588 } | |
5589 } | |
5590 /* for processed > 0 */ | |
5591 else | |
5592 { | |
5593 /* Keep count of how many buffers we have | |
5594 * to queue so we can return the value | |
5595 */ | |
5596 retval++; | |
5597 /* | |
5598 fprintf(stderr, "NOT_EOF, about to Queue more data for filled max_queue (%d)\n", | |
5599 ALmixer_Channel_List[i].almixer_data->max_queue_buffers); | |
5600 */ | |
5601 alSourceQueueBuffers( | |
5602 ALmixer_Channel_List[i].alsource, | |
5603 1, &unqueued_buffer_id); | |
5604 if((error = alGetError()) != AL_NO_ERROR) | |
5605 { | |
5606 ALmixer_SetError("Could not QueueBuffer: %s", | |
2
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|
5607 alGetString(error) ); |
0 | 5608 error_flag--; |
5609 continue; | |
5610 } | |
5611 /* This is part of the hideous Nvidia workaround. In order to figure out | |
5612 * which buffer to show during callbacks (for things like | |
5613 * o-scopes), I must keep a copy of the buffers that are queued in my own | |
5614 * data structure. This code will be called only if | |
5615 * "access_data" was set, indicated by whether the queue is NULL. | |
5616 */ | |
5617 if(ALmixer_Channel_List[i].almixer_data->circular_buffer_queue != NULL) | |
5618 { | |
2
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|
5619 ALuint queue_ret_flag; |
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diff
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|
5620 // fprintf(stderr, "56e: CircularQueue_PushBack.\n"); |
0 | 5621 queue_ret_flag = CircularQueueUnsignedInt_PushBack( |
5622 ALmixer_Channel_List[i].almixer_data->circular_buffer_queue, | |
5623 unqueued_buffer_id | |
5624 ); | |
5625 if(0 == queue_ret_flag) | |
5626 { | |
5627 fprintf(stderr, "56bSerious internal error: CircularQueue could not push into queue.\n"); | |
5628 ALmixer_SetError("Serious internal error: CircularQueue failed to push into queue"); | |
5629 } | |
2
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|
5630 #if 0 |
0 | 5631 else |
5632 { | |
5633 CircularQueueUnsignedInt_Print(ALmixer_Channel_List[i].almixer_data->circular_buffer_queue); | |
5634 } | |
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|
5635 #endif |
0 | 5636 } |
5637 } | |
5638 /* If we used an available buffer queue, | |
5639 * then we need to update the number of them in use | |
5640 */ | |
5641 if(ALmixer_Channel_List[i].almixer_data->num_buffers_in_use | |
5642 < ALmixer_Channel_List[i].almixer_data->max_queue_buffers) | |
5643 { | |
5644 /* Increment the number of buffers in use */ | |
5645 ALmixer_Channel_List[i].almixer_data->num_buffers_in_use++; | |
5646 } | |
5647 /* Might want to check state */ | |
5648 /* In case the playback stopped, | |
5649 * we need to resume */ | |
1 | 5650 #if 1 |
5651 /* Try not refetching the state here because I'm getting a duplicate | |
5652 buffer playback (hiccup) */ | |
0 | 5653 alGetSourcei( |
5654 ALmixer_Channel_List[i].alsource, | |
5655 AL_SOURCE_STATE, &state | |
5656 ); | |
1 | 5657 if((error = alGetError()) != AL_NO_ERROR) |
5658 { | |
5659 fprintf(stderr, "57bTesting error: %s\n", | |
2
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|
5660 alGetString(error)); |
1 | 5661 } |
5662 /* Get the number of buffers processed | |
5663 */ | |
5664 alGetSourcei( | |
5665 ALmixer_Channel_List[i].alsource, | |
5666 AL_BUFFERS_PROCESSED, | |
5667 &buffers_processed | |
5668 ); | |
5669 if((error = alGetError()) != AL_NO_ERROR) | |
5670 { | |
5671 fprintf(stderr, "57cError, Can't get buffers_processed: %s\n", | |
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|
5672 alGetString(error)); |
1 | 5673 } |
5674 #endif | |
0 | 5675 if(AL_STOPPED == state) |
5676 { | |
1 | 5677 /* |
0 | 5678 fprintf(stderr, "Resuming in not eof\n"); |
1 | 5679 */ |
5680 /* Okay, here's another lately discovered problem: | |
5681 * I can't find it in the spec, but for at least some of the | |
5682 * implementations, if I call play on a stopped source that | |
5683 * has processed buffers, all those buffers get marked as unprocessed | |
5684 * on alSourcePlay. So if I had a queue of 25 with 24 of the buffers | |
5685 * processed, on resume, the earlier 24 buffers will get replayed, | |
5686 * causing a "hiccup" like sound in the playback. | |
5687 * To avoid this, I must unqueue all processed buffers before | |
5688 * calling play. But to complicate things, I need to worry about resyncing | |
5689 * the circular queue with this since I designed this thing | |
5690 * with some correlation between the two. However, I might | |
5691 * have already handled this, so I will try writing this code without | |
5692 * syncing for now. | |
5693 * There is currently an assumption that a buffer | |
5694 * was queued above so I actually have something | |
5695 * to play. | |
5696 */ | |
5697 ALint temp_count; | |
5698 /* | |
5699 fprintf(stderr, "STOPPED2, need to clear processed, status is:\n"); | |
5700 PrintQueueStatus(ALmixer_Channel_List[i].alsource); | |
5701 */ | |
5702 | |
5703 for(temp_count=0; temp_count<buffers_processed; temp_count++) | |
5704 { | |
5705 alSourceUnqueueBuffers( | |
5706 ALmixer_Channel_List[i].alsource, | |
5707 1, &unqueued_buffer_id | |
5708 ); | |
5709 if((error = alGetError()) != AL_NO_ERROR) | |
5710 { | |
5711 fprintf(stderr, "58aTesting error: %s\n", | |
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|
5712 alGetString(error)); |
1 | 5713 error_flag--; |
5714 } | |
5715 } | |
5716 /* | |
5717 fprintf(stderr, "After unqueue clear...:\n"); | |
5718 PrintQueueStatus(ALmixer_Channel_List[i].alsource); | |
5719 */ | |
5720 | |
5721 alSourcePlay(ALmixer_Channel_List[i].alsource); | |
5722 if((error = alGetError()) != AL_NO_ERROR) | |
5723 { | |
5724 fprintf(stderr, "55Tbesting 8rror: %s\n", | |
2
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|
5725 alGetString(error)); |
1 | 5726 } |
0 | 5727 } |
5728 continue; | |
5729 } /* END if( ! eof) */ | |
5730 /* We have hit EOF in the SDL_Sound sample and there | |
5731 * are no more loops. However, there may still be | |
5732 * buffers in the OpenAL queue which still need to | |
5733 * be played out. The following body of code will | |
5734 * determine if play is still happening or | |
5735 * initiate the stop/cleanup sequenece. | |
5736 */ | |
5737 else | |
5738 { | |
5739 /* Let's continue to remove the used up | |
5740 * buffers as they come in. */ | |
5741 if(buffers_processed > 0) | |
5742 { | |
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|
5743 ALint temp_count; |
0 | 5744 /* Do as a for-loop because I don't want |
5745 * to have to create an array for the | |
5746 * unqueued_buffer_id's | |
5747 */ | |
5748 for(temp_count=0; temp_count<buffers_processed; temp_count++) | |
5749 { | |
5750 fprintf(stderr, "unqueuing remainder, %d\n", temp_count); | |
5751 alSourceUnqueueBuffers( | |
5752 ALmixer_Channel_List[i].alsource, | |
5753 1, &unqueued_buffer_id | |
5754 ); | |
5755 if((error = alGetError()) != AL_NO_ERROR) | |
5756 { | |
5757 fprintf(stderr, "59Testing error: %s\n", | |
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|
5758 alGetString(error)); |
0 | 5759 } |
5760 } | |
5761 fprintf(stderr, "done unqueuing remainder for this loop, %d\n", temp_count); | |
5762 | |
5763 /* Need to update counts since we removed everything. | |
5764 * If we don't update the counts here, we end up in the | |
5765 * "Shouldn't be here section, but maybe it's okay due to race conditions" | |
5766 */ | |
5767 alGetSourcei( | |
5768 ALmixer_Channel_List[i].alsource, | |
5769 AL_BUFFERS_QUEUED, &buffers_still_queued | |
5770 ); | |
5771 if((error = alGetError()) != AL_NO_ERROR) | |
5772 { | |
5773 fprintf(stderr, "5100Testing error: %s\n", | |
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|
5774 alGetString(error)); |
0 | 5775 } |
5776 /* Get the number of buffers processed | |
5777 * so we know if we need to refill | |
5778 */ | |
5779 alGetSourcei( | |
5780 ALmixer_Channel_List[i].alsource, | |
5781 AL_BUFFERS_PROCESSED, &buffers_processed | |
5782 ); | |
5783 if((error = alGetError()) != AL_NO_ERROR) | |
5784 { | |
5785 fprintf(stderr, "5200Testing error: %s\n", | |
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|
5786 alGetString(error)); |
0 | 5787 } |
5788 } | |
5789 | |
5790 | |
5791 /* Else if buffers_processed == 0 | |
5792 * and buffers_still_queued == 0. | |
5793 * then we check to see if the source | |
5794 * is still playing. Quit if stopped | |
5795 * We shouldn't need to worry about | |
5796 * looping because that should have | |
5797 * been handled above. | |
5798 */ | |
5799 if(0 == buffers_still_queued) | |
5800 { | |
5801 /* Make sure playback has stopped before | |
5802 * we shutdown. | |
5803 */ | |
5804 alGetSourcei( | |
5805 ALmixer_Channel_List[i].alsource, | |
5806 AL_SOURCE_STATE, &state | |
5807 ); | |
5808 if((error = alGetError()) != AL_NO_ERROR) | |
5809 { | |
5810 fprintf(stderr, "60Testing error: %s\n", | |
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|
5811 alGetString(error)); |
0 | 5812 } |
5813 if(AL_STOPPED == state) | |
5814 { | |
5815 ALmixer_Channel_List[i].almixer_data->num_buffers_in_use = 0; | |
5816 /* Playback has ended. | |
5817 * Loop if necessary, or launch callback | |
5818 * and clear channel (or clear channel and | |
5819 * then launch callback?) | |
5820 */ | |
5821 Clean_Channel(i); | |
5822 /* Subtract counter */ | |
5823 Is_Playing_global--; | |
5824 | |
5825 /* Launch callback */ | |
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|
5826 Invoke_Channel_Done_Callback(i, AL_TRUE); |
0 | 5827 |
5828 /* We're done for this loop. | |
5829 * Go to next channel | |
5830 */ | |
5831 continue; | |
5832 } | |
5833 } /* End end-playback */ | |
5834 else | |
5835 { | |
5836 /* Need to run out buffer */ | |
5837 #if 1 | |
5838 /* Might want to check state */ | |
5839 /* In case the playback stopped, | |
5840 * we need to resume */ | |
5841 alGetSourcei( | |
5842 ALmixer_Channel_List[i].alsource, | |
5843 AL_SOURCE_STATE, &state | |
5844 ); | |
5845 if((error = alGetError()) != AL_NO_ERROR) | |
5846 { | |
5847 fprintf(stderr, "61Testing error: %s\n", | |
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5848 alGetString(error)); |
0 | 5849 } |
5850 if(AL_STOPPED == state) | |
5851 { | |
5852 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); | |
1 | 5853 /* |
0 | 5854 PrintQueueStatus(ALmixer_Channel_List[i].alsource); |
1 | 5855 */ |
0 | 5856 /* Rather than force unqueuing the buffer, let's see if |
5857 * setting the buffer to none works (the OpenAL 1.0 | |
5858 * Reference Annotation suggests this should work). | |
5859 */ | |
5860 alSourcei(ALmixer_Channel_List[i].alsource, | |
5861 AL_BUFFER, AL_NONE); | |
5862 /* | |
5863 PrintQueueStatus(ALmixer_Channel_List[i].alsource); | |
5864 */ | |
5865 /* This doesn't work because in some cases, I think | |
5866 * it causes the sound to be replayed | |
5867 */ | |
5868 /* | |
5869 fprintf(stderr, "Resuming in eof (trying to run out buffers\n"); | |
5870 alSourcePlay(ALmixer_Channel_List[i].alsource); | |
5871 */ | |
5872 } | |
5873 #endif | |
5874 } /* End trap section */ | |
5875 } /* End POST-EOF use-up buffer section */ | |
5876 } /* END Streamed section */ | |
5877 } /* END channel in use */ | |
5878 } /* END for-loop for each channel */ | |
5879 | |
5880 #ifdef ENABLE_ALMIXER_ALC_SYNC | |
5881 alcProcessContext(alcGetCurrentContext()); | |
5882 if((error = alGetError()) != AL_NO_ERROR) | |
5883 { | |
5884 fprintf(stderr, "62Testing error: %s\n", | |
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5885 alGetString(error)); |
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5886 } |
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5887 #endif |
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5888 |
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5889 #ifdef ENABLE_ALMIXER_THREADS |
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5890 SDL_UnlockMutex(s_simpleLock); |
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5891 #endif |
0 | 5892 /* Return the number of errors */ |
5893 if(error_flag < 0) | |
5894 { | |
5895 return error_flag; | |
5896 } | |
5897 /* Return the number of buffers that were queued */ | |
5898 return retval; | |
5899 } | |
5900 | |
5901 #ifdef ENABLE_PARANOID_SIGNEDNESS_CHECK | |
5902 /* This is only here so we can call SDL_OpenAudio() */ | |
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5903 static void my_dummy_audio_callback(void* userdata, ALbyte* stream, int len) |
0 | 5904 { |
5905 } | |
5906 #endif | |
5907 | |
5908 | |
5909 | |
5910 | |
5911 #ifdef ENABLE_ALMIXER_THREADS | |
5912 /* We might need threads. We | |
5913 * must constantly poll OpenAL to find out | |
5914 * if sound is being streamed, if play has | |
5915 * ended, etc. Without threads, this must | |
5916 * be explicitly done by the user. | |
5917 * We could try to do it for them if we | |
5918 * finish the threads. | |
5919 */ | |
5920 | |
5921 static int Stream_Data_Thread_Callback(void* data) | |
5922 { | |
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5923 ALint retval; |
0 | 5924 |
5925 while(ALmixer_Initialized) | |
5926 { | |
5927 retval = Update_ALmixer(data); | |
5928 /* 0 means that nothing needed updating and | |
5929 * the function returned quickly | |
5930 */ | |
5931 if(0 == retval) | |
5932 { | |
5933 /* Let's be nice and make the thread sleep since | |
5934 * little work was done in update | |
5935 */ | |
5936 /* Make sure times are multiples of 10 | |
5937 * for optimal performance and accuracy in Linux | |
5938 */ | |
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5939 ALmixer_Delay(10); |
0 | 5940 } |
5941 else | |
5942 { | |
5943 /* should I also be sleeping/yielding here? */ | |
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5944 ALmixer_Delay(0); |
0 | 5945 } |
5946 } | |
5947 fprintf(stderr, "Thread is closing\n"); | |
5948 return 0; | |
5949 } | |
5950 #endif /* End of ENABLE_ALMIXER_THREADS */ | |
5951 | |
5952 | |
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5953 /* SDL/SDL_mixer returns -1 on error and 0 on success. |
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5954 * I actually prefer false/true conventions (SDL_Sound/OpenAL/GL) |
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5955 * so SDL_mixer porting people beware. |
0 | 5956 * Warning: SDL_QuitSubSystem(SDL_INIT_AUDIO) is called which |
5957 * means the SDL audio system will be disabled. It will not | |
5958 * be restored (in case SDL is not actually being used) so | |
5959 * the user will need to restart it if they need it after | |
5960 * OpenAL shuts down. | |
5961 */ | |
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5962 ALboolean ALmixer_Init(ALuint frequency, ALint num_sources, ALuint refresh) |
0 | 5963 { |
5964 ALCdevice* dev; | |
5965 ALCcontext* context; | |
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5966 ALint i; |
0 | 5967 ALenum error; |
5968 ALuint* source; | |
5969 | |
5970 #ifdef USING_LOKI_AL_DIST | |
5971 /* The Loki dist requires that I set both the | |
5972 * device and context frequency values separately | |
5973 */ | |
5974 /* Hope this won't overflow */ | |
5975 char device_string[256]; | |
5976 #endif | |
5977 | |
5978 /* (Venting frustration) Damn it! Nobody bothered | |
5979 * documenting how you're supposed to use an attribute | |
5980 * list. In fact, the not even the Loki test program | |
5981 * writers seem to know because they use it inconsistently. | |
5982 * For example, how do you terminate that attribute list? | |
5983 * The Loki test code does it 3 different ways. They | |
5984 * set the last value to 0, or they set it to ALC_INVALID, | |
5985 * or they set two final values: ALC_INVALID, 0 | |
5986 * In Loki, 0 and ALC_INVALID happen to be the same, | |
5987 * but with Creative Labs ALC_INVALID is -1. | |
5988 * So something's going to break. Loki's source | |
5989 * code says to terminate with ALC_INVALID. But I | |
5990 * don't know if that's really true, or it happens | |
5991 * to be a coinicidence because it's defined to 0. | |
5992 * Creative provides no source code, so I can't look at how | |
5993 * they terminate it. | |
5994 * So this is really, really ticking me off... | |
5995 * For now, I'm going to use ALC_INVALID. | |
5996 * (Update...after further review of the API spec, | |
5997 * it seems that a NULL terminated string is the correct | |
5998 * termination value to use, so 0 it is.) | |
5999 */ | |
6000 #if 0 | |
6001 ALint attrlist[] = { | |
6002 ALC_FREQUENCY, ALMIXER_DEFAULT_FREQUENCY, | |
6003 /* Don't know anything about these values. | |
6004 * Trust defaults? */ | |
6005 /* Supposed to be the refresh rate in Hz. | |
6006 * I think 15-120 are supposed to be good | |
6007 * values. Though I haven't gotten any effect except | |
6008 * for one strange instance on a Mac. But it was | |
6009 * unrepeatable. | |
6010 */ | |
6011 #if 0 | |
6012 ALC_REFRESH, 15, | |
6013 #endif | |
6014 /* Sync requires a alcProcessContext() call | |
6015 * for every cycle. By default, this is | |
6016 * not used and the value is AL_FALSE | |
6017 * because it will probably perform | |
6018 * pretty badly for me. | |
6019 */ | |
6020 #ifdef ENABLE_ALMIXER_ALC_SYNC | |
6021 ALC_SYNC, AL_TRUE, | |
6022 #else | |
6023 ALC_SYNC, AL_FALSE, | |
6024 #endif | |
6025 /* Looking at the API spec, it implies | |
6026 * that the list be a NULL terminated string | |
6027 * so it's probably not safe to use ALC_INVALID | |
6028 */ | |
6029 /* | |
6030 ALC_INVALID }; | |
6031 */ | |
6032 '\0'}; | |
6033 #endif | |
6034 /* Redo: I'm going to allow ALC_REFRESH to be set. | |
6035 * However, if no value is specified, I don't | |
6036 * want it in the list so I can get the OpenAL defaults | |
6037 */ | |
6038 ALint attrlist[7]; | |
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6039 ALsizei current_attrlist_index = 0; |
0 | 6040 |
6041 #ifdef ENABLE_PARANOID_SIGNEDNESS_CHECK | |
6042 /* More problems: I'm getting bit by endian/signedness issues on | |
6043 * different platforms. I can find the endianess easily enough, | |
6044 * but I don't know how to determine what the correct signedness | |
6045 * is (if such a thing exists). I do know that if I try using | |
6046 * unsigned on OSX with an originally signed sample, I get | |
6047 * distortion. However, I don't have any native unsigned samples | |
6048 * to test. But I'm assuming that the platform must be in the | |
6049 * correct signedness no matter what. | |
6050 * I can either assume everybody is signed, or I can try to | |
6051 * determine the value. If I try to determine the values, | |
6052 * I think my only ability to figure it out will be to open | |
6053 * SDL_Audio, and read what the obtained settings were. | |
6054 * Then shutdown everything. However, I don't even know how | |
6055 * reliable this is. | |
6056 * Update: I think I resolved the issues...forgot to update | |
6057 * these comments when it happened. I should check the revision control | |
6058 * log... Anyway, I think the issue was partly related to me not | |
6059 * doing something correctly with the AudioInfo or some kind | |
6060 * of stupid endian bug in my code, and weirdness ensued. Looking at the | |
6061 * revision control, I think I might have assumed that SDL_Sound would | |
6062 * do the right thing with a NULL AudioInfo, but I was incorrect, | |
6063 * and had to fill one out myself. | |
6064 */ | |
6065 SDL_AudioSpec desired; | |
6066 SDL_AudioSpec obtained; | |
6067 #endif | |
6068 | |
6069 | |
6070 /* Make sure ALmixer isn't already initialized */ | |
6071 if(ALmixer_Initialized) | |
6072 { | |
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6073 return AL_FALSE; |
0 | 6074 } |
6075 #ifdef USING_LOKI_AL_DIST | |
6076 fprintf(stderr, "Found Loki dist\n"); | |
6077 #elif defined(USING_CREATIVE_AL_DIST) | |
6078 fprintf(stderr, "Found Creative dist\n"); | |
6079 | |
6080 #elif defined(USING_NVIDIA_AL_DIST) | |
6081 fprintf(stderr, "Found Nvidia dist\n"); | |
6082 #endif | |
6083 | |
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6084 #ifdef ALMIXER_COMPILE_WITHOUT_SDL |
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6085 ALmixer_InitTime(); |
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6086 |
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6087 /* Note: The pool may have been created on previous Init's */ |
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6088 /* I leave the pool allocated allocated in case the user wants |
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6089 * to read the pool in case of a failure (such as in this function). |
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6090 * This is not actually a leak. |
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6091 */ |
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6092 if(NULL == s_ALmixerErrorPool) |
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6093 { |
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6094 s_ALmixerErrorPool = TError_CreateErrorPool(); |
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6095 } |
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6096 if(NULL == s_ALmixerErrorPool) |
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6097 { |
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6098 return AL_FALSE; |
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6099 } |
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6100 fprintf(stderr, "tError Test0\n"); |
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6101 ALmixer_SetError("Initing (and testing SetError)"); |
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6102 fprintf(stderr, "tError Test1: %s\n", ALmixer_GetError()); |
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6103 fprintf(stderr, "tError Test2: %s\n", ALmixer_GetError()); |
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6104 #endif |
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6105 |
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6106 |
0 | 6107 /* Set the defaults */ |
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6108 /* |
0 | 6109 attrlist[0] = ALC_FREQUENCY; |
6110 attrlist[1] = ALMIXER_DEFAULT_FREQUENCY; | |
6111 attrlist[2] = ALC_SYNC; | |
6112 #ifdef ENABLE_ALMIXER_ALC_SYNC | |
6113 attrlist[3] = ALC_TRUE; | |
6114 #else | |
6115 attrlist[3] = ALC_FALSE; | |
6116 #endif | |
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6117 */ |
0 | 6118 /* Set frequency value if it is not 0 */ |
6119 if(0 != frequency) | |
6120 { | |
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6121 attrlist[current_attrlist_index] = ALC_FREQUENCY; |
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6122 current_attrlist_index++; |
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6123 attrlist[current_attrlist_index] = (ALint)frequency; |
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6124 current_attrlist_index++; |
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6125 } |
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6126 |
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6127 #ifdef ENABLE_ALMIXER_ALC_SYNC |
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6128 attrlist[current_attrlist_index] = ALC_SYNC; |
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6129 current_attrlist_index++; |
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6130 attrlist[current_attrlist_index] = ALC_TRUE; |
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6131 current_attrlist_index++; |
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6132 #endif |
0 | 6133 |
6134 /* If the user specifies a refresh value, | |
6135 * make room for it | |
6136 */ | |
6137 if(0 != refresh) | |
6138 { | |
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6139 attrlist[current_attrlist_index] = (ALint)ALC_REFRESH; |
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6140 current_attrlist_index++; |
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6141 attrlist[current_attrlist_index] = refresh; |
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6142 current_attrlist_index++; |
0 | 6143 } |
6144 | |
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6145 /* End attribute list */ |
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6146 attrlist[current_attrlist_index] = '\0'; |
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6147 |
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6148 |
0 | 6149 /* Initialize SDL_Sound */ |
6150 if(! Sound_Init() ) | |
6151 { | |
6152 ALmixer_SetError(Sound_GetError()); | |
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6153 return AL_FALSE; |
0 | 6154 } |
6155 #ifdef ENABLE_PARANOID_SIGNEDNESS_CHECK | |
6156 /* Here is the paranoid check that opens | |
6157 * SDL audio in an attempt to find the correct | |
6158 * system values. | |
6159 */ | |
6160 /* Doesn't have to be the actual value I think | |
6161 * (as long as it doesn't influence format, in | |
6162 * which case I'm probably screwed anyway because OpenAL | |
6163 * may easily choose to do something else). | |
6164 */ | |
6165 desired.freq = 44100; | |
6166 desired.channels = 2; | |
6167 desired.format = AUDIO_S16SYS; | |
6168 desired.callback = my_dummy_audio_callback; | |
6169 if(SDL_OpenAudio(&desired, &obtained) >= 0) | |
6170 { | |
6171 SIGN_TYPE_16BIT_FORMAT = obtained.format; | |
6172 /* Now to get really paranoid, we should probably | |
6173 * also assume that the 8bit format is also the | |
6174 * same sign type and set that value | |
6175 */ | |
6176 if(AUDIO_S16SYS == obtained.format) | |
6177 { | |
6178 SIGN_TYPE_8BIT_FORMAT = AUDIO_S8; | |
6179 } | |
6180 /* Should be AUDIO_U16SYS */ | |
6181 else | |
6182 { | |
6183 SIGN_TYPE_8BIT_FORMAT = AUDIO_U8; | |
6184 } | |
6185 SDL_CloseAudio(); | |
6186 fprintf(stderr, "Obtained format = %d", obtained.format); | |
6187 } | |
6188 else | |
6189 { | |
6190 /* Well, I guess I'm in trouble. I guess it's my best guess | |
6191 */ | |
6192 SIGN_TYPE_16_BIT_FORMAT = AUDIO_S16SYS; | |
6193 SIGN_TYPE_8_BIT_FORMAT = AUDIO_S8; | |
6194 } | |
6195 #endif | |
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6196 |
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6197 #ifndef ALMIXER_COMPILE_WITHOUT_SDL |
0 | 6198 /* Weirdness: It seems that SDL_Init(SDL_INIT_AUDIO) |
6199 * causes OpenAL and SMPEG to conflict. For some reason | |
6200 * if SDL_Init on audio is active, then all the SMPEG | |
6201 * decoded sound comes out silent. Unfortunately, | |
6202 * Sound_Init() invokes SDL_Init on audio. I'm | |
6203 * not sure why it actually needs it... | |
6204 * But we'll attempt to disable it here after the | |
6205 * SDL_Sound::Init call and hope it doesn't break SDL_Sound. | |
6206 */ | |
6207 SDL_QuitSubSystem(SDL_INIT_AUDIO); | |
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6208 #endif |
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6209 |
0 | 6210 /* I'm told NULL will call the default string |
6211 * and hopefully do the right thing for each platform | |
6212 */ | |
6213 /* | |
6214 dev = alcOpenDevice( NULL ); | |
6215 */ | |
6216 /* Now I'm told I need to set both the device and context | |
6217 * to have the same sampling rate, so I must pass a string | |
6218 * to OpenDevice(). I don't know how portable these strings are. | |
6219 * I don't even know if the format for strings is | |
6220 * compatible | |
6221 * From the testattrib.c in the Loki test section | |
6222 * dev = alcOpenDevice( (const ALubyte *) "'((sampling-rate 22050))" ); | |
6223 */ | |
6224 | |
6225 #ifdef USING_LOKI_AL_DIST | |
6226 sprintf(device_string, "'((sampling-rate %d))", attrlist[1]); | |
6227 dev = alcOpenDevice( (const ALubyte *) device_string ); | |
6228 #else | |
6229 dev = alcOpenDevice( NULL ); | |
6230 #endif | |
6231 fprintf(stderr,"sampling-rate is %d\n", attrlist[1]); | |
6232 if(NULL == dev) | |
6233 { | |
6234 ALmixer_SetError("Cannot open sound device for OpenAL"); | |
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6235 return AL_FALSE; |
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6236 } |
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6237 |
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6238 #ifdef __APPLE__ |
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6239 /* The ALC_FREQUENCY attribute is ignored with Apple's implementation. */ |
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6240 /* This extension must be called before the context is created. */ |
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6241 if(0 != frequency) |
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6242 { |
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6243 Internal_alcMacOSXMixerOutputRate((ALdouble)frequency); |
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6244 } |
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6245 ALmixer_Frequency_global = (ALuint)Internal_alcMacOSXGetMixerOutputRate(); |
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6246 fprintf(stderr, "Internal_alcMacOSXMixerOutputRate is: %lf", Internal_alcMacOSXGetMixerOutputRate()); |
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6247 #endif |
0 | 6248 |
6249 context = alcCreateContext(dev, attrlist); | |
6250 if(NULL == context) | |
6251 { | |
6252 ALmixer_SetError("Cannot create a context OpenAL"); | |
6253 alcCloseDevice(dev); | |
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6254 return AL_FALSE; |
0 | 6255 } |
6256 fprintf(stderr, "Context checking...\n"); | |
6257 | |
6258 | |
6259 /* Hmmm, OSX is returning 1 on alcMakeContextCurrent, | |
6260 * but ALC_NO_ERROR is defined to ALC_FALSE. | |
6261 * According to Garin Hiebert, this is actually an inconsistency | |
6262 * in the Loki version. The function should return a boolean. | |
6263 * instead of ALC_NO_ERROR. Garin suggested I check via | |
6264 * alcGetError(). | |
6265 */ | |
6266 /* clear the error */ | |
6267 alcGetError(dev); | |
6268 alcMakeContextCurrent(context); | |
6269 | |
6270 error = alcGetError(dev); | |
6271 if( (ALC_NO_ERROR != error) ) | |
6272 { | |
6273 ALmixer_SetError("Could not MakeContextCurrent"); | |
6274 alcDestroyContext(context); | |
6275 alcCloseDevice(dev); | |
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6276 return AL_FALSE; |
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6277 } |
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6278 |
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6279 /* It looks like OpenAL won't let us ask it what |
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6280 * the set frequency is, so we need to save our |
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6281 * own copy. Yuck. |
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6282 * Update: J. Valenzuela just updated the Loki |
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6283 * dist (2003/01/02) to handle this. |
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6284 * The demo is in testattrib.c. |
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6285 */ |
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6286 /* |
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6287 ALmixer_Frequency_global = frequency; |
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6288 */ |
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6289 #ifndef __APPLE__ |
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6290 alcGetIntegerv(dev, ALC_FREQUENCY, 1, &ALmixer_Frequency_global); |
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6291 fprintf(stderr, "alcGetIntegerv ALC_FREQUENCY is: %d", ALmixer_Frequency_global); |
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6292 #endif |
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6293 |
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6294 |
0 | 6295 #if 0 |
6296 /* OSX is failing on alcMakeContextCurrent(). Try checking it first? */ | |
6297 if(alcGetCurrentContext() != context) | |
6298 { | |
6299 /* Hmmm, OSX is returning 1 on alcMakeContextCurrent, | |
6300 * but ALC_NO_ERROR is defined to ALC_FALSE. | |
6301 * I think this is a bug in the OpenAL implementation. | |
6302 */ | |
6303 fprintf(stderr,"alcMakeContextCurrent returns %d\n", alcMakeContextCurrent(context)); | |
6304 | |
6305 fprintf(stderr, "Making context current\n"); | |
6306 #ifndef __APPLE__ | |
6307 if(alcMakeContextCurrent(context) != ALC_NO_ERROR) | |
6308 #else | |
6309 if(!alcMakeContextCurrent(context)) | |
6310 #endif | |
6311 { | |
6312 ALmixer_SetError("Could not MakeContextCurrent"); | |
6313 alcDestroyContext(context); | |
6314 alcCloseDevice(dev); | |
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6315 return AL_FALSE; |
0 | 6316 } |
6317 } | |
6318 #endif | |
6319 | |
6320 | |
6321 /* #endif */ | |
6322 fprintf(stderr, "done Context\n"); | |
6323 /* Saw this in the README with the OS X OpenAL distribution. | |
6324 * It looked interesting and simple, so I thought I might | |
6325 * try it out. | |
6326 * ***** ALC_CONVERT_DATA_UPON_LOADING | |
6327 * This extension allows the caller to tell OpenAL to preconvert to the native Core | |
6328 * Audio format, the audio data passed to the | |
6329 * library with the alBufferData() call. Preconverting the audio data, reduces CPU | |
6330 * usage by removing an audio data conversion | |
6331 * (per source) at render timem at the expense of a larger memory footprint. | |
6332 * | |
6333 * This feature is toggled on/off by using the alDisable() & alEnable() APIs. This | |
6334 * setting will be applied to all subsequent | |
6335 * calls to alBufferData(). | |
6336 */ | |
6337 #ifdef __APPLE__ | |
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6338 /* |
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6339 #if (TARGET_OS_IPHONE == 1) || (TARGET_IPHONE_SIMULATOR == 1) |
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6340 |
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6341 #else |
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6342 #endif |
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6343 */ |
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6344 ALenum convert_data_enum = alcGetEnumValue(dev, "ALC_MAC_OSX_CONVERT_DATA_UPON_LOADING"); |
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6345 fprintf(stderr, "ALC_MAC_OSX_CONVERT_DATA_UPON_LOADING=0x%x", convert_data_enum); |
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6346 if(0 != convert_data_enum) |
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6347 { |
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6348 alEnable(convert_data_enum); |
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6349 } |
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6350 if( (AL_NO_ERROR != alGetError()) ) |
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6351 { |
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6352 ALmixer_SetError("ALC_MAC_OSX_CONVERT_DATA_UPON_LOADING attempted but failed"); |
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6353 } |
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6354 |
0 | 6355 #endif |
6356 | |
6357 | |
6358 | |
6359 | |
6360 ALmixer_Initialized = 1; | |
6361 | |
6362 if(num_sources <= 0) | |
6363 { | |
6364 Number_of_Channels_global = ALMIXER_DEFAULT_NUM_CHANNELS; | |
6365 } | |
6366 else | |
6367 { | |
6368 Number_of_Channels_global = num_sources; | |
6369 } | |
6370 Number_of_Reserve_Channels_global = 0; | |
6371 Is_Playing_global = 0; | |
6372 /* Set to Null in case system quit and was reinitialized */ | |
6373 Channel_Done_Callback = NULL; | |
6374 Channel_Done_Callback_Userdata = NULL; | |
6375 Channel_Data_Callback = NULL; | |
1 | 6376 Channel_Data_Callback_Userdata = NULL; |
0 | 6377 |
6378 /* Allocate memory for the list of channels */ | |
6379 ALmixer_Channel_List = (struct ALmixer_Channel*) malloc(Number_of_Channels_global * sizeof(struct ALmixer_Channel)); | |
6380 if(NULL == ALmixer_Channel_List) | |
6381 { | |
6382 ALmixer_SetError("Out of Memory for Channel List"); | |
6383 alcDestroyContext(context); | |
6384 alcCloseDevice(dev); | |
6385 ALmixer_Initialized = 0; | |
6386 Number_of_Channels_global = 0; | |
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6387 return AL_FALSE; |
0 | 6388 } |
6389 | |
6390 /* Allocate memory for the list of sources that map to the channels */ | |
6391 Source_Map_List = (Source_Map*) malloc(Number_of_Channels_global * sizeof(Source_Map)); | |
6392 if(NULL == Source_Map_List) | |
6393 { | |
6394 ALmixer_SetError("Out of Memory for Source Map List"); | |
6395 free(ALmixer_Channel_List); | |
6396 alcDestroyContext(context); | |
6397 alcCloseDevice(dev); | |
6398 ALmixer_Initialized = 0; | |
6399 Number_of_Channels_global = 0; | |
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6400 return AL_FALSE; |
0 | 6401 } |
6402 | |
6403 /* Create array that will hold the sources */ | |
6404 source = (ALuint*)malloc(Number_of_Channels_global * sizeof(ALuint)); | |
6405 if(NULL == source) | |
6406 { | |
6407 ALmixer_SetError("Out of Memory for sources"); | |
6408 free(Source_Map_List); | |
6409 free(ALmixer_Channel_List); | |
6410 alcDestroyContext(context); | |
6411 alcCloseDevice(dev); | |
6412 ALmixer_Initialized = 0; | |
6413 Number_of_Channels_global = 0; | |
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6414 return AL_FALSE; |
0 | 6415 } |
6416 | |
6417 /* Clear the error state */ | |
6418 alGetError(); | |
6419 /* Generate the OpenAL sources */ | |
6420 alGenSources(Number_of_Channels_global, source); | |
6421 if( (error=alGetError()) != AL_NO_ERROR) | |
6422 { | |
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6423 ALmixer_SetError("Couldn't generate sources: %s\n", alGetString(error)); |
0 | 6424 free(ALmixer_Channel_List); |
6425 free(Source_Map_List); | |
6426 alcDestroyContext(context); | |
6427 alcCloseDevice(dev); | |
6428 ALmixer_Initialized = 0; | |
6429 Number_of_Channels_global = 0; | |
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6430 return AL_FALSE; |
0 | 6431 } |
6432 | |
6433 /* Initialize each channel and associate one source to one channel */ | |
6434 for(i=0; i<Number_of_Channels_global; i++) | |
6435 { | |
6436 if(0 == source[i]) | |
6437 { | |
6438 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"); | |
6439 } | |
6440 | |
6441 Init_Channel(i); | |
6442 /* Keeping the source allocation out of the Init function | |
6443 * in case I want to reuse the Init | |
6444 * function for resetting data | |
6445 */ | |
6446 ALmixer_Channel_List[i].alsource = source[i]; | |
6447 /* Now also keep a copy of the source to channel mapping | |
6448 * in case we need to look up a channel from the source | |
6449 * instead of a source from a channel | |
6450 */ | |
6451 Source_Map_List[i].source = source[i]; | |
6452 Source_Map_List[i].channel = i; | |
6453 /* Clean the channel because there are some things that need to | |
6454 * be done that can't happen until the source is set | |
6455 */ | |
6456 Clean_Channel(i); | |
6457 } | |
6458 | |
6459 /* The Source_Map_List must be sorted by source for binary searches | |
6460 */ | |
6461 qsort(Source_Map_List, Number_of_Channels_global, sizeof(Source_Map), Compare_Source_Map); | |
6462 | |
6463 fprintf(stderr, "Sorted Source_Map_List is:\n"); | |
6464 for(i=0; i<Number_of_Channels_global; i++) | |
6465 { | |
6466 fprintf(stderr, "Source: %d, Channel: %d\n", Source_Map_List[i].source, Source_Map_List[i].channel); | |
6467 } | |
6468 fprintf(stderr, "\n"); | |
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6469 ALmixer_OutputDecoders(); |
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6470 |
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6471 #ifdef ENABLE_ALMIXER_THREADS |
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6472 s_simpleLock = SDL_CreateMutex(); |
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6473 if(NULL == s_simpleLock) |
0 | 6474 { |
6475 /* SDL sets the error message already? */ | |
6476 free(source); | |
6477 free(ALmixer_Channel_List); | |
6478 free(Source_Map_List); | |
6479 alcDestroyContext(context); | |
6480 alcCloseDevice(dev); | |
6481 ALmixer_Initialized = 0; | |
6482 Number_of_Channels_global = 0; | |
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6483 return AL_FALSE; |
0 | 6484 } |
6485 | |
6486 | |
6487 Stream_Thread_global = SDL_CreateThread(Stream_Data_Thread_Callback, NULL); | |
6488 if(NULL == Stream_Thread_global) | |
6489 { | |
6490 /* SDL sets the error message already? */ | |
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6491 SDL_DestroyMutex(s_simpleLock); |
0 | 6492 free(source); |
6493 free(ALmixer_Channel_List); | |
6494 free(Source_Map_List); | |
6495 alcDestroyContext(context); | |
6496 alcCloseDevice(dev); | |
6497 ALmixer_Initialized = 0; | |
6498 Number_of_Channels_global = 0; | |
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6499 return AL_FALSE; |
0 | 6500 } |
6501 | |
6502 fprintf(stderr, "Using threads\n"); | |
6503 #endif /* End of ENABLE_ALMIXER_THREADS */ | |
6504 | |
6505 /* We don't need this array any more because all the sources | |
6506 * are connected to channels | |
6507 */ | |
6508 free(source); | |
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6509 return AL_TRUE; |
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6510 } |
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6511 |
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6512 |
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6513 ALboolean ALmixer_InitContext(ALuint frequency, ALuint refresh) |
0 | 6514 { |
6515 ALCdevice* dev; | |
6516 ALCcontext* context; | |
6517 ALCenum error; | |
6518 | |
6519 #ifdef USING_LOKI_AL_DIST | |
6520 /* The Loki dist requires that I set both the | |
6521 * device and context frequency values separately | |
6522 */ | |
6523 /* Hope this won't overflow */ | |
6524 char device_string[256]; | |
6525 #endif | |
6526 | |
6527 /* (Venting frustration) Damn it! Nobody bothered | |
6528 * documenting how you're supposed to use an attribute | |
6529 * list. In fact, the not even the Loki test program | |
6530 * writers seem to know because they use it inconsistently. | |
6531 * For example, how do you terminate that attribute list? | |
6532 * The Loki test code does it 3 different ways. They | |
6533 * set the last value to 0, or they set it to ALC_INVALID, | |
6534 * or they set two final values: ALC_INVALID, 0 | |
6535 * In Loki, 0 and ALC_INVALID happen to be the same, | |
6536 * but with Creative Labs ALC_INVALID is -1. | |
6537 * So something's going to break. Loki's source | |
6538 * code says to terminate with ALC_INVALID. But I | |
6539 * don't know if that's really true, or it happens | |
6540 * to be a coinicidence because it's defined to 0. | |
6541 * Creative provides no source code, so I can't look at how | |
6542 * they terminate it. | |
6543 * So this is really, really ticking me off... | |
6544 * For now, I'm going to use ALC_INVALID. | |
6545 * (Update...after further review of the API spec, | |
6546 * it seems that a NULL terminated string is the correct | |
6547 * termination value to use, so 0 it is.) | |
6548 */ | |
6549 #if 0 | |
6550 ALint attrlist[] = { | |
6551 ALC_FREQUENCY, ALMIXER_DEFAULT_FREQUENCY, | |
6552 /* Don't know anything about these values. | |
6553 * Trust defaults? */ | |
6554 /* Supposed to be the refresh rate in Hz. | |
6555 * I think 15-120 are supposed to be good | |
6556 * values. Though I haven't gotten any effect except | |
6557 * for one strange instance on a Mac. But it was | |
6558 * unrepeatable. | |
6559 */ | |
6560 #if 0 | |
6561 ALC_REFRESH, 15, | |
6562 #endif | |
6563 /* Sync requires a alcProcessContext() call | |
6564 * for every cycle. By default, this is | |
6565 * not used and the value is AL_FALSE | |
6566 * because it will probably perform | |
6567 * pretty badly for me. | |
6568 */ | |
6569 #ifdef ENABLE_ALMIXER_ALC_SYNC | |
6570 ALC_SYNC, AL_TRUE, | |
6571 #else | |
6572 ALC_SYNC, AL_FALSE, | |
6573 #endif | |
6574 /* Looking at the API spec, it implies | |
6575 * that the list be a NULL terminated string | |
6576 * so it's probably not safe to use ALC_INVALID | |
6577 */ | |
6578 /* | |
6579 ALC_INVALID }; | |
6580 */ | |
6581 '\0'}; | |
6582 #endif | |
6583 /* Redo: I'm going to allow ALC_REFRESH to be set. | |
6584 * However, if no value is specified, I don't | |
6585 * want it in the list so I can get the OpenAL defaults | |
6586 */ | |
6587 ALint attrlist[7]; | |
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6588 ALsizei current_attrlist_index = 0; |
0 | 6589 |
6590 #ifdef ENABLE_PARANOID_SIGNEDNESS_CHECK | |
6591 /* More problems: I'm getting bit by endian/signedness issues on | |
6592 * different platforms. I can find the endianess easily enough, | |
6593 * but I don't know how to determine what the correct signedness | |
6594 * is (if such a thing exists). I do know that if I try using | |
6595 * unsigned on OSX with an originally signed sample, I get | |
6596 * distortion. However, I don't have any native unsigned samples | |
6597 * to test. But I'm assuming that the platform must be in the | |
6598 * correct signedness no matter what. | |
6599 * I can either assume everybody is signed, or I can try to | |
6600 * determine the value. If I try to determine the values, | |
6601 * I think my only ability to figure it out will be to open | |
6602 * SDL_Audio, and read what the obtained settings were. | |
6603 * Then shutdown everything. However, I don't even know how | |
6604 * reliable this is. | |
6605 * Update: I think I resolved the issues...forgot to update | |
6606 * these comments when it happened. I should check the revision control | |
6607 * log... Anyway, I think the issue was partly related to me not | |
6608 * doing something correctly with the AudioInfo or some kind | |
6609 * of stupid endian bug in my code, and weirdness ensued. Looking at the | |
6610 * revision control, I think I might have assumed that SDL_Sound would | |
6611 * do the right thing with a NULL AudioInfo, but I was incorrect, | |
6612 * and had to fill one out myself. | |
6613 */ | |
6614 SDL_AudioSpec desired; | |
6615 SDL_AudioSpec obtained; | |
6616 #endif | |
6617 | |
6618 | |
6619 | |
6620 | |
6621 /* Make sure ALmixer isn't already initialized */ | |
6622 if(ALmixer_Initialized) | |
6623 { | |
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6624 return AL_FALSE; |
0 | 6625 } |
6626 #ifdef USING_LOKI_AL_DIST | |
6627 fprintf(stderr, "Found Loki dist\n"); | |
6628 #elif defined(USING_CREATIVE_AL_DIST) | |
6629 fprintf(stderr, "Found Creative dist\n"); | |
6630 | |
6631 #elif defined(USING_NVIDIA_AL_DIST) | |
6632 fprintf(stderr, "Found Nvidia dist\n"); | |
6633 #endif | |
6634 | |
6635 /* Set the defaults */ | |
6636 attrlist[0] = ALC_FREQUENCY; | |
6637 attrlist[1] = ALMIXER_DEFAULT_FREQUENCY; | |
6638 attrlist[2] = ALC_SYNC; | |
6639 #ifdef ENABLE_ALMIXER_ALC_SYNC | |
6640 attrlist[3] = ALC_TRUE; | |
6641 #else | |
6642 attrlist[3] = ALC_FALSE; | |
6643 #endif | |
6644 /* Set frequency value if it is not 0 */ | |
6645 if(0 != frequency) | |
6646 { | |
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6647 attrlist[current_attrlist_index] = ALC_FREQUENCY; |
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6648 current_attrlist_index++; |
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6649 attrlist[current_attrlist_index] = (ALint)frequency; |
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6650 current_attrlist_index++; |
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6651 } |
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6652 |
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6653 #ifdef ENABLE_ALMIXER_ALC_SYNC |
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6654 attrlist[current_attrlist_index] = ALC_SYNC; |
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6655 current_attrlist_index++; |
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6656 attrlist[current_attrlist_index] = ALC_TRUE; |
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6657 current_attrlist_index++; |
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6658 #endif |
0 | 6659 |
6660 /* If the user specifies a refresh value, | |
6661 * make room for it | |
6662 */ | |
6663 if(0 != refresh) | |
6664 { | |
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6665 attrlist[current_attrlist_index] = (ALint)ALC_REFRESH; |
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6666 current_attrlist_index++; |
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6667 attrlist[current_attrlist_index] = refresh; |
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6668 current_attrlist_index++; |
0 | 6669 } |
6670 | |
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6671 /* End attribute list */ |
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6672 attrlist[current_attrlist_index] = '\0'; |
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6673 |
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6674 |
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6675 |
0 | 6676 /* Initialize SDL_Sound */ |
6677 if(! Sound_Init() ) | |
6678 { | |
6679 ALmixer_SetError(Sound_GetError()); | |
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6680 return AL_FALSE; |
0 | 6681 } |
6682 #ifdef ENABLE_PARANOID_SIGNEDNESS_CHECK | |
6683 /* Here is the paranoid check that opens | |
6684 * SDL audio in an attempt to find the correct | |
6685 * system values. | |
6686 */ | |
6687 /* Doesn't have to be the actual value I think | |
6688 * (as long as it doesn't influence format, in | |
6689 * which case I'm probably screwed anyway because OpenAL | |
6690 * may easily choose to do something else). | |
6691 */ | |
6692 desired.freq = 44100; | |
6693 desired.channels = 2; | |
6694 desired.format = AUDIO_S16SYS; | |
6695 desired.callback = my_dummy_audio_callback; | |
6696 if(SDL_OpenAudio(&desired, &obtained) >= 0) | |
6697 { | |
6698 SIGN_TYPE_16BIT_FORMAT = obtained.format; | |
6699 /* Now to get really paranoid, we should probably | |
6700 * also assume that the 8bit format is also the | |
6701 * same sign type and set that value | |
6702 */ | |
6703 if(AUDIO_S16SYS == obtained.format) | |
6704 { | |
6705 SIGN_TYPE_8BIT_FORMAT = AUDIO_S8; | |
6706 } | |
6707 /* Should be AUDIO_U16SYS */ | |
6708 else | |
6709 { | |
6710 SIGN_TYPE_8BIT_FORMAT = AUDIO_U8; | |
6711 } | |
6712 SDL_CloseAudio(); | |
6713 fprintf(stderr, "Obtained format = %d", obtained.format); | |
6714 } | |
6715 else | |
6716 { | |
6717 /* Well, I guess I'm in trouble. I guess it's my best guess | |
6718 */ | |
6719 SIGN_TYPE_16_BIT_FORMAT = AUDIO_S16SYS; | |
6720 SIGN_TYPE_8_BIT_FORMAT = AUDIO_S8; | |
6721 } | |
6722 #endif | |
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6723 |
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6724 #ifndef ALMIXER_COMPILE_WITHOUT_SDL |
0 | 6725 /* Weirdness: It seems that SDL_Init(SDL_INIT_AUDIO) |
6726 * causes OpenAL and SMPEG to conflict. For some reason | |
6727 * if SDL_Init on audio is active, then all the SMPEG | |
6728 * decoded sound comes out silent. Unfortunately, | |
6729 * Sound_Init() invokes SDL_Init on audio. I'm | |
6730 * not sure why it actually needs it... | |
6731 * But we'll attempt to disable it here after the | |
6732 * SDL_Sound::Init call and hope it doesn't break SDL_Sound. | |
6733 */ | |
6734 SDL_QuitSubSystem(SDL_INIT_AUDIO); | |
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6735 #endif |
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6736 |
0 | 6737 /* I'm told NULL will call the default string |
6738 * and hopefully do the right thing for each platform | |
6739 */ | |
6740 /* | |
6741 dev = alcOpenDevice( NULL ); | |
6742 */ | |
6743 /* Now I'm told I need to set both the device and context | |
6744 * to have the same sampling rate, so I must pass a string | |
6745 * to OpenDevice(). I don't know how portable these strings are. | |
6746 * I don't even know if the format for strings is | |
6747 * compatible | |
6748 * From the testattrib.c in the Loki test section | |
6749 * dev = alcOpenDevice( (const ALubyte *) "'((sampling-rate 22050))" ); | |
6750 */ | |
6751 | |
6752 #ifdef USING_LOKI_AL_DIST | |
6753 sprintf(device_string, "'((sampling-rate %d))", attrlist[1]); | |
6754 dev = alcOpenDevice( (const ALubyte *) device_string ); | |
6755 #else | |
6756 dev = alcOpenDevice( NULL ); | |
6757 #endif | |
6758 fprintf(stderr,"sampling-rate is %d\n", attrlist[1]); | |
6759 if(NULL == dev) | |
6760 { | |
6761 ALmixer_SetError("Cannot open sound device for OpenAL"); | |
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6762 return AL_FALSE; |
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6763 } |
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6764 |
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6765 #ifdef __APPLE__ |
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6766 /* The ALC_FREQUENCY attribute is ignored with Apple's implementation. */ |
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6767 /* This extension must be called before the context is created. */ |
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6768 if(0 != frequency) |
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6769 { |
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6770 Internal_alcMacOSXMixerOutputRate((ALdouble)frequency); |
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6771 } |
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6772 ALmixer_Frequency_global = (ALuint)Internal_alcMacOSXGetMixerOutputRate(); |
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6773 fprintf(stderr, "Internal_alcMacOSXMixerOutputRate is: %lf", Internal_alcMacOSXGetMixerOutputRate()); |
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6774 #endif |
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6775 |
0 | 6776 |
6777 context = alcCreateContext(dev, attrlist); | |
6778 if(NULL == context) | |
6779 { | |
6780 ALmixer_SetError("Cannot create a context OpenAL"); | |
6781 alcCloseDevice(dev); | |
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6782 return AL_FALSE; |
0 | 6783 } |
6784 | |
6785 | |
6786 /* Hmmm, OSX is returning 1 on alcMakeContextCurrent, | |
6787 * but ALC_NO_ERROR is defined to ALC_FALSE. | |
6788 * According to Garin Hiebert, this is actually an inconsistency | |
6789 * in the Loki version. The function should return a boolean. | |
6790 * instead of ALC_NO_ERROR. Garin suggested I check via | |
6791 * alcGetError(). | |
6792 */ | |
6793 /* clear the error */ | |
6794 alcGetError(dev); | |
6795 alcMakeContextCurrent(context); | |
6796 | |
6797 error = alcGetError(dev); | |
6798 if( (ALC_NO_ERROR != error) ) | |
6799 { | |
6800 ALmixer_SetError("Could not MakeContextCurrent"); | |
6801 alcDestroyContext(context); | |
6802 alcCloseDevice(dev); | |
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6803 return AL_FALSE; |
0 | 6804 } |
6805 | |
6806 | |
6807 #if 0 | |
6808 /* OSX is failing on alcMakeContextCurrent(). Try checking it first? */ | |
6809 if(alcGetCurrentContext() != context) | |
6810 { | |
6811 /* Hmmm, OSX is returning 1 on alcMakeContextCurrent, | |
6812 * but ALC_NO_ERROR is defined to ALC_FALSE. | |
6813 * I think this is a bug in the OpenAL implementation. | |
6814 */ | |
6815 fprintf(stderr,"alcMakeContextCurrent returns %d\n", alcMakeContextCurrent(context)); | |
6816 | |
6817 fprintf(stderr, "Making context current\n"); | |
6818 #ifndef __APPLE__ | |
6819 if(alcMakeContextCurrent(context) != ALC_NO_ERROR) | |
6820 #else | |
6821 if(!alcMakeContextCurrent(context)) | |
6822 #endif | |
6823 { | |
6824 ALmixer_SetError("Could not MakeContextCurrent"); | |
6825 alcDestroyContext(context); | |
6826 alcCloseDevice(dev); | |
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6827 return AL_FALSE; |
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6828 } |
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6829 |
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6830 } |
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6831 #endif |
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6832 |
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6833 /* It looks like OpenAL won't let us ask it what |
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6834 * the set frequency is, so we need to save our |
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6835 * own copy. Yuck. |
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6836 * Update: J. Valenzuela just updated the Loki |
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6837 * dist (2003/01/02) to handle this. |
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6838 * The demo is in testattrib.c. |
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6839 */ |
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6840 #ifndef __APPLE__ |
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6841 alcGetIntegerv(dev, ALC_FREQUENCY, 1, &ALmixer_Frequency_global); |
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6842 fprintf(stderr, "alcGetIntegerv ALC_FREQUENCY is: %d", ALmixer_Frequency_global); |
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6843 #endif |
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6844 |
0 | 6845 |
6846 fprintf(stderr, "done Context\n"); | |
6847 | |
6848 /* Saw this in the README with the OS X OpenAL distribution. | |
6849 * It looked interesting and simple, so I thought I might | |
6850 * try it out. | |
6851 * ***** ALC_CONVERT_DATA_UPON_LOADING | |
6852 * This extension allows the caller to tell OpenAL to preconvert to the native Core | |
6853 * Audio format, the audio data passed to the | |
6854 * library with the alBufferData() call. Preconverting the audio data, reduces CPU | |
6855 * usage by removing an audio data conversion | |
6856 * (per source) at render timem at the expense of a larger memory footprint. | |
6857 * | |
6858 * This feature is toggled on/off by using the alDisable() & alEnable() APIs. This | |
6859 * setting will be applied to all subsequent | |
6860 * calls to alBufferData(). | |
6861 */ | |
6862 #ifdef __APPLE__ | |
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6863 /* |
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6864 #if (TARGET_OS_IPHONE == 1) || (TARGET_IPHONE_SIMULATOR == 1) |
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6865 |
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6866 #else |
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6867 #endif |
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6868 */ |
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6869 ALenum convert_data_enum = alcGetEnumValue(dev, "ALC_MAC_OSX_CONVERT_DATA_UPON_LOADING"); |
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6870 fprintf(stderr, "ALC_MAC_OSX_CONVERT_DATA_UPON_LOADING=0x%x", convert_data_enum); |
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6871 if(0 != convert_data_enum) |
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6872 { |
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6873 alEnable(convert_data_enum); |
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6874 } |
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6875 if( (AL_NO_ERROR != alGetError()) ) |
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6876 { |
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6877 ALmixer_SetError("ALC_MAC_OSX_CONVERT_DATA_UPON_LOADING attempted but failed"); |
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6878 } |
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6879 #endif |
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6880 |
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6881 return AL_TRUE; |
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6882 } |
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6883 |
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6884 |
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6885 ALboolean ALmixer_InitMixer(ALint num_sources) |
0 | 6886 { |
2
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6887 ALint i; |
0 | 6888 ALenum error; |
6889 ALuint* source; | |
6890 | |
6891 | |
6892 ALmixer_Initialized = 1; | |
6893 | |
2
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6894 |
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6895 #ifdef ALMIXER_COMPILE_WITHOUT_SDL |
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6896 ALmixer_InitTime(); |
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6897 |
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6898 /* Note: The pool may have been created on previous Init's */ |
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6899 /* I leave the pool allocated allocated in case the user wants |
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6900 * to read the pool in case of a failure (such as in this function). |
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6901 * This is not actually a leak. |
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6902 */ |
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6903 if(NULL == s_ALmixerErrorPool) |
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6904 { |
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6905 s_ALmixerErrorPool = TError_CreateErrorPool(); |
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6906 } |
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6907 if(NULL == s_ALmixerErrorPool) |
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6908 { |
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6909 return AL_FALSE; |
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6910 } |
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6911 /* |
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6912 fprintf(stderr, "tError Test0\n"); |
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6913 ALmixer_SetError("Initing (and testing SetError)"); |
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6914 fprintf(stderr, "tError Test1: %s\n", ALmixer_GetError()); |
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6915 fprintf(stderr, "tError Test2: %s\n", ALmixer_GetError()); |
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6916 */ |
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6917 #endif |
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6918 |
0 | 6919 if(num_sources <= 0) |
6920 { | |
6921 Number_of_Channels_global = ALMIXER_DEFAULT_NUM_CHANNELS; | |
6922 } | |
6923 else | |
6924 { | |
6925 Number_of_Channels_global = num_sources; | |
6926 } | |
6927 Number_of_Reserve_Channels_global = 0; | |
6928 Is_Playing_global = 0; | |
6929 /* Set to Null in case system quit and was reinitialized */ | |
6930 Channel_Done_Callback = NULL; | |
6931 Channel_Done_Callback_Userdata = NULL; | |
6932 Channel_Data_Callback = NULL; | |
1 | 6933 Channel_Data_Callback_Userdata = NULL; |
0 | 6934 |
6935 /* Allocate memory for the list of channels */ | |
6936 ALmixer_Channel_List = (struct ALmixer_Channel*) malloc(Number_of_Channels_global * sizeof(struct ALmixer_Channel)); | |
6937 if(NULL == ALmixer_Channel_List) | |
6938 { | |
6939 ALmixer_SetError("Out of Memory for Channel List"); | |
6940 ALmixer_Initialized = 0; | |
6941 Number_of_Channels_global = 0; | |
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6942 return AL_FALSE; |
0 | 6943 } |
6944 | |
6945 /* Allocate memory for the list of sources that map to the channels */ | |
6946 Source_Map_List = (Source_Map*) malloc(Number_of_Channels_global * sizeof(Source_Map)); | |
6947 if(NULL == Source_Map_List) | |
6948 { | |
6949 ALmixer_SetError("Out of Memory for Source Map List"); | |
6950 free(ALmixer_Channel_List); | |
6951 ALmixer_Initialized = 0; | |
6952 Number_of_Channels_global = 0; | |
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6953 return AL_FALSE; |
0 | 6954 } |
6955 | |
6956 /* Create array that will hold the sources */ | |
6957 source = (ALuint*)malloc(Number_of_Channels_global * sizeof(ALuint)); | |
6958 if(NULL == source) | |
6959 { | |
6960 ALmixer_SetError("Out of Memory for sources"); | |
6961 free(Source_Map_List); | |
6962 free(ALmixer_Channel_List); | |
6963 ALmixer_Initialized = 0; | |
6964 Number_of_Channels_global = 0; | |
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6965 return AL_FALSE; |
0 | 6966 } |
6967 | |
6968 /* Clear the error state */ | |
6969 alGetError(); | |
6970 /* Generate the OpenAL sources */ | |
6971 alGenSources(Number_of_Channels_global, source); | |
6972 if( (error=alGetError()) != AL_NO_ERROR) | |
6973 { | |
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6974 ALmixer_SetError("Couldn't generate sources: %s\n", alGetString(error)); |
0 | 6975 free(ALmixer_Channel_List); |
6976 free(Source_Map_List); | |
6977 ALmixer_Initialized = 0; | |
6978 Number_of_Channels_global = 0; | |
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6979 return AL_FALSE; |
0 | 6980 } |
6981 | |
6982 /* Initialize each channel and associate one source to one channel */ | |
6983 for(i=0; i<Number_of_Channels_global; i++) | |
6984 { | |
6985 Init_Channel(i); | |
6986 /* Keeping the source allocation out of the Init function | |
6987 * in case I want to reuse the Init | |
6988 * function for resetting data | |
6989 */ | |
6990 ALmixer_Channel_List[i].alsource = source[i]; | |
6991 /* Now also keep a copy of the source to channel mapping | |
6992 * in case we need to look up a channel from the source | |
6993 * instead of a source from a channel | |
6994 */ | |
6995 Source_Map_List[i].source = source[i]; | |
6996 Source_Map_List[i].channel = i; | |
6997 /* Clean the channel because there are some things that need to | |
6998 * be done that can't happen until the source is set | |
6999 */ | |
7000 Clean_Channel(i); | |
7001 } | |
7002 | |
7003 /* The Source_Map_List must be sorted by source for binary searches | |
7004 */ | |
7005 qsort(Source_Map_List, Number_of_Channels_global, sizeof(Source_Map), Compare_Source_Map); | |
7006 | |
7007 fprintf(stderr, "Sorted Source_Map_List is:\n"); | |
7008 for(i=0; i<Number_of_Channels_global; i++) | |
7009 { | |
7010 fprintf(stderr, "Source: %d, Channel: %d\n", Source_Map_List[i].source, Source_Map_List[i].channel); | |
7011 } | |
7012 fprintf(stderr, "\n"); | |
7013 | |
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7014 |
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7015 |
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7016 #ifdef ENABLE_ALMIXER_THREADS |
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7017 s_simpleLock = SDL_CreateMutex(); |
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7018 if(NULL == s_simpleLock) |
0 | 7019 { |
7020 /* SDL sets the error message already? */ | |
7021 free(source); | |
7022 free(ALmixer_Channel_List); | |
7023 free(Source_Map_List); | |
7024 ALmixer_Initialized = 0; | |
7025 Number_of_Channels_global = 0; | |
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7026 return AL_FALSE; |
0 | 7027 } |
7028 | |
7029 | |
7030 Stream_Thread_global = SDL_CreateThread(Stream_Data_Thread_Callback, NULL); | |
7031 if(NULL == Stream_Thread_global) | |
7032 { | |
7033 /* SDL sets the error message already? */ | |
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7034 SDL_DestroyMutex(s_simpleLock); |
0 | 7035 free(source); |
7036 free(ALmixer_Channel_List); | |
7037 free(Source_Map_List); | |
7038 ALmixer_Initialized = 0; | |
7039 Number_of_Channels_global = 0; | |
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7040 return AL_FALSE; |
0 | 7041 } |
7042 | |
7043 fprintf(stderr, "Using threads\n"); | |
7044 #endif /* End of ENABLE_ALMIXER_THREADS */ | |
7045 | |
7046 /* We don't need this array any more because all the sources | |
7047 * are connected to channels | |
7048 */ | |
7049 free(source); | |
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7050 return AL_TRUE; |
0 | 7051 } |
7052 | |
7053 | |
7054 | |
7055 /* Keep the return value void to allow easy use with | |
7056 * atexit() | |
7057 */ | |
7058 void ALmixer_Quit() | |
7059 { | |
7060 ALCcontext* context; | |
7061 ALCdevice* dev; | |
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7062 ALint i; |
0 | 7063 |
7064 if( ! ALmixer_Initialized) | |
7065 { | |
7066 return; | |
7067 } | |
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7068 #ifdef ENABLE_ALMIXER_THREADS |
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7069 SDL_LockMutex(s_simpleLock); |
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7070 #endif |
0 | 7071 /* Shutdown everything before closing context */ |
7072 fprintf(stderr, "Halting channels\n"); | |
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7073 Internal_HaltChannel(-1, AL_FALSE); |
0 | 7074 |
7075 /* This flag will cause the thread to terminate */ | |
7076 ALmixer_Initialized = 0; | |
7077 #ifdef ENABLE_ALMIXER_THREADS | |
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7078 SDL_UnlockMutex(s_simpleLock); |
0 | 7079 fprintf(stderr, "Closing thread\n"); |
7080 SDL_WaitThread(Stream_Thread_global, NULL); | |
7081 | |
7082 fprintf(stderr, "Destroying mutex\n"); | |
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7083 SDL_DestroyMutex(s_simpleLock); |
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7084 #endif |
0 | 7085 |
7086 fprintf(stderr, "Deleting OpenAL sources\n"); | |
7087 /* Delete all the OpenAL sources */ | |
7088 for(i=0; i<Number_of_Channels_global; i++) | |
7089 { | |
7090 fprintf(stderr, "Deleting OpenAL source: %d\n", ALmixer_Channel_List[i].alsource); | |
7091 alDeleteSources(1, &ALmixer_Channel_List[i].alsource); | |
7092 } | |
7093 /* Delete all the channels */ | |
7094 free(ALmixer_Channel_List); | |
7095 free(Source_Map_List); | |
7096 | |
7097 /* Reset the Number_of_Channels just in case somebody | |
7098 * tries using a ALmixer function. | |
7099 * I probably should put "Initialized" checks everywhere, | |
7100 * but I'm too lazy at the moment. | |
7101 */ | |
7102 Number_of_Channels_global = 0; | |
7103 | |
7104 context = alcGetCurrentContext(); | |
7105 if(NULL == context) | |
7106 { | |
7107 return; | |
7108 } | |
7109 /* Need to get the device before I close the context */ | |
7110 dev = alcGetContextsDevice(context); | |
7111 alcDestroyContext(context); | |
7112 | |
7113 if(NULL == dev) | |
7114 { | |
7115 return; | |
7116 } | |
7117 alcCloseDevice(dev); | |
7118 | |
7119 Sound_Quit(); | |
7120 | |
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7121 #ifdef ALMIXER_COMPILE_WITHOUT_SDL |
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7122 /* Remember: ALmixer_SetError/GetError calls will not work while this is gone. */ |
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7123 TError_FreeErrorPool(s_ALmixerErrorPool); |
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7124 s_ALmixerErrorPool = NULL; |
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7125 #endif |
0 | 7126 return; |
7127 } | |
7128 | |
2
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7129 ALboolean ALmixer_IsInitialized() |
0 | 7130 { |
7131 return ALmixer_Initialized; | |
7132 } | |
7133 | |
2
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7134 ALuint ALmixer_GetFrequency() |
0 | 7135 { |
7136 return ALmixer_Frequency_global; | |
7137 } | |
7138 | |
2
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7139 const ALmixer_version* ALmixer_GetLinkedVersion() |
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7140 { |
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7141 static ALmixer_version linked_mixver; |
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7142 ALMIXER_GET_COMPILED_VERSION(&linked_mixver); |
0 | 7143 return(&linked_mixver); |
7144 } | |
7145 | |
2
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7146 #ifdef ALMIXER_COMPILE_WITHOUT_SDL |
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7147 |
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7148 const char* ALmixer_GetError() |
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7149 { |
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7150 const char* error_string = NULL; |
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7151 if(NULL == s_ALmixerErrorPool) |
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7152 { |
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7153 return "Error: You should not call ALmixer_GetError while ALmixer is not initialized"; |
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7154 } |
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7155 error_string = TError_GetLastErrorStr(s_ALmixerErrorPool); |
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7156 /* SDL returns empty strings instead of NULL */ |
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7157 if(NULL == error_string) |
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7158 { |
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7159 return ""; |
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7160 } |
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7161 else |
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7162 { |
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7163 return error_string; |
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7164 } |
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7165 } |
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7166 |
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7167 void ALmixer_SetError(const char* err_str, ...) |
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7168 { |
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7169 if(NULL == s_ALmixerErrorPool) |
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7170 { |
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7171 fprintf(stderr, "Error: You should not call ALmixer_SetError while ALmixer is not initialized\n"); |
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7172 return; |
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7173 } |
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7174 va_list argp; |
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7175 va_start(argp, err_str); |
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7176 // SDL_SetError which I'm emulating has no number parameter. |
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7177 TError_SetErrorv(s_ALmixerErrorPool, 1, err_str, argp); |
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7178 va_end(argp); |
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7179 } |
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7180 |
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7181 #endif |
0 | 7182 |
7183 | |
7184 | |
7185 | |
7186 #if 0 | |
2
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7187 void ALmixer_OutputAttributes() |
0 | 7188 { |
7189 ALint num_flags = 0; | |
7190 ALint* flags = 0; | |
7191 int i; | |
7192 ALCdevice* dev = alcGetContextsDevice( alcGetCurrentContext() ); | |
7193 | |
7194 | |
7195 printf("custom context\n"); | |
7196 | |
7197 alcGetIntegerv(dev, ALC_ATTRIBUTES_SIZE, | |
7198 sizeof num_flags, &num_flags ); | |
7199 | |
7200 printf("Number of Flags: %d\n", num_flags); | |
7201 | |
7202 if(num_flags) | |
7203 { | |
7204 flags = malloc(sizeof(num_flags) * sizeof(ALint)); | |
7205 | |
7206 alcGetIntegerv(dev, ALC_ALL_ATTRIBUTES, | |
7207 sizeof num_flags * sizeof(ALint), | |
7208 flags ); | |
7209 } | |
7210 for(i = 0; i < num_flags-1; i += 2) | |
7211 { | |
7212 printf("key 0x%x : value %d\n", | |
7213 flags[i], flags[i+1]); | |
7214 } | |
7215 free(flags); | |
7216 } | |
7217 #endif | |
7218 | |
7219 | |
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7220 void ALmixer_OutputDecoders() |
0 | 7221 { |
7222 Sound_Version sound_compile_version; | |
7223 Sound_Version sound_link_version; | |
7224 | |
7225 const Sound_DecoderInfo **rc = Sound_AvailableDecoders(); | |
7226 const Sound_DecoderInfo **i; | |
7227 const char **ext; | |
7228 FILE* stream = stdout; | |
7229 | |
7230 | |
7231 fprintf(stream, "SDL_sound Information:\n"); | |
7232 | |
7233 SOUND_VERSION(&sound_compile_version); | |
7234 fprintf(stream, "\tCompiled with SDL_sound version: %d.%d.%d\n", | |
7235 sound_compile_version.major, | |
7236 sound_compile_version.minor, | |
7237 sound_compile_version.patch); | |
7238 | |
7239 Sound_GetLinkedVersion(&sound_link_version); | |
7240 fprintf(stream, "\tRunning (linked) with SDL_sound version: %d.%d.%d\n", | |
7241 sound_link_version.major, | |
7242 sound_link_version.minor, | |
7243 sound_link_version.patch); | |
7244 | |
7245 fprintf(stream, "Supported sound formats:\n"); | |
7246 if (rc == NULL) | |
7247 fprintf(stream, " * Apparently, NONE!\n"); | |
7248 else | |
7249 { | |
7250 for (i = rc; *i != NULL; i++) | |
7251 { | |
7252 fprintf(stream, " * %s\n", (*i)->description); | |
7253 | |
7254 for (ext = (*i)->extensions; *ext != NULL; ext++) | |
7255 fprintf(stream, " File extension \"%s\"\n", *ext); | |
7256 | |
7257 fprintf(stream, " Written by %s.\n %s\n\n", | |
7258 (*i)->author, (*i)->url); | |
7259 } /* for */ | |
7260 } /* else */ | |
7261 | |
7262 fprintf(stream, "\n"); | |
7263 } | |
7264 | |
2
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7265 void ALmixer_OutputOpenALInfo() |
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7266 { |
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7267 ALmixer_version mixer_compile_version; |
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7268 const ALmixer_version * mixer_link_version=ALmixer_GetLinkedVersion(); |
0 | 7269 FILE* stream = stdout; |
7270 | |
7271 fprintf(stream, "OpenAL Information:\n"); | |
7272 fprintf(stream, "\tAL_VENDOR: %s\n", alGetString( AL_VENDOR ) ); | |
7273 fprintf(stream, "\tAL_VERSION: %s\n", alGetString( AL_VERSION ) ); | |
7274 fprintf(stream, "\tAL_RENDERER: %s\n", alGetString( AL_RENDERER ) ); | |
7275 fprintf(stream, "\tAL_EXTENSIONS: %s\n", alGetString( AL_EXTENSIONS ) ); | |
7276 | |
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7277 ALMIXER_GET_COMPILED_VERSION(&mixer_compile_version); |
0 | 7278 fprintf(stream, "\nSDL_ALmixer Information:\n"); |
7279 fprintf(stream, "\tCompiled with SDL_ALmixer version: %d.%d.%d\n", | |
7280 mixer_compile_version.major, | |
7281 mixer_compile_version.minor, | |
7282 mixer_compile_version.patch); | |
7283 | |
7284 fprintf(stream, "\tRunning (linked) with SDL_ALmixer version: %d.%d.%d\n", | |
7285 mixer_link_version->major, | |
7286 mixer_link_version->minor, | |
7287 mixer_link_version->patch); | |
7288 | |
7289 fprintf(stream, "\tCompile flags: "); | |
7290 #ifdef ENABLE_LOKI_QUEUE_FIX_HACK | |
7291 fprintf(stream, "ENABLE_LOKI_QUEUE_FIX_HACK "); | |
7292 #endif | |
7293 #ifdef ENABLE_ALMIXER_THREADS | |
7294 fprintf(stream, "ENABLE_ALMIXER_THREADS "); | |
7295 #endif | |
7296 #ifdef ENABLE_ALC_SYNC | |
7297 fprintf(stream, "ENABLE_ALC_SYNC "); | |
7298 #endif | |
7299 fprintf(stream, "\n"); | |
7300 } | |
7301 | |
7302 | |
2
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7303 ALint ALmixer_AllocateChannels(ALint numchans) |
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7304 { |
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7305 ALint retval; |
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7306 #ifdef ENABLE_ALMIXER_THREADS |
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7307 SDL_LockMutex(s_simpleLock); |
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7308 #endif |
0 | 7309 retval = Internal_AllocateChannels(numchans); |
2
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7310 #ifdef ENABLE_ALMIXER_THREADS |
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7311 SDL_UnlockMutex(s_simpleLock); |
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7312 #endif |
0 | 7313 return retval; |
7314 } | |
7315 | |
7316 | |
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7317 ALint ALmixer_ReserveChannels(ALint num) |
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7318 { |
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7319 ALint retval; |
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7320 #ifdef ENABLE_ALMIXER_THREADS |
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7321 SDL_LockMutex(s_simpleLock); |
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7322 #endif |
0 | 7323 retval = Internal_ReserveChannels(num); |
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7324 #ifdef ENABLE_ALMIXER_THREADS |
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7325 SDL_UnlockMutex(s_simpleLock); |
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7326 #endif |
0 | 7327 return retval; |
7328 } | |
7329 | |
7330 | |
7331 | |
7332 | |
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7333 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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7334 { |
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7335 ALuint bytes_decoded; |
0 | 7336 ALmixer_Data* ret_data; |
7337 ALenum error; | |
7338 | |
7339 /* Allocate memory */ | |
7340 ret_data = (ALmixer_Data *)malloc(sizeof(ALmixer_Data)); | |
7341 if (NULL == ret_data) | |
7342 { | |
7343 ALmixer_SetError("Out of memory"); | |
7344 return(NULL); | |
7345 } | |
7346 | |
7347 /* Initialize the data fields */ | |
7348 | |
7349 /* Set the Sound_Sample pointer */ | |
7350 ret_data->sample = sample; | |
7351 | |
7352 /* Flag the data to note that it is not in use */ | |
7353 ret_data->in_use = 0; | |
7354 | |
7355 /* Initialize remaining flags */ | |
7356 ret_data->total_time = -1; | |
7357 ret_data->eof = 0; | |
7358 | |
7359 /* Just initialize */ | |
7360 ret_data->num_buffers_in_use = 0; | |
7361 | |
7362 /* Just initialize */ | |
7363 ret_data->total_bytes = 0; | |
7364 | |
7365 /* Just initialize */ | |
7366 ret_data->loaded_bytes = 0; | |
7367 | |
7368 /* Set the max queue buffers (minimum must be 2) */ | |
7369 if(max_queue_buffers < 2) | |
7370 { | |
7371 max_queue_buffers = ALMIXER_DEFAULT_QUEUE_BUFFERS; | |
7372 } | |
7373 ret_data->max_queue_buffers = max_queue_buffers; | |
7374 /* Set up the start up buffers */ | |
7375 if(0 == num_startup_buffers) | |
7376 { | |
7377 num_startup_buffers = ALMIXER_DEFAULT_STARTUP_BUFFERS; | |
7378 } | |
7379 /* Make sure start up buffers is less or equal to max_queue_buffers */ | |
7380 if(num_startup_buffers > max_queue_buffers) | |
7381 { | |
7382 num_startup_buffers = max_queue_buffers; | |
7383 } | |
7384 ret_data->num_startup_buffers = num_startup_buffers; | |
7385 | |
7386 ret_data->buffer_map_list = NULL; | |
7387 ret_data->current_buffer = 0; | |
7388 | |
7389 ret_data->circular_buffer_queue = NULL; | |
7390 | |
7391 /* Now decode and load the data into a data chunk */ | |
7392 /* Different cases for Streamed and Predecoded | |
7393 * Streamed might turn into a predecoded if buffersize | |
7394 * is large enough */ | |
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7395 if(AL_FALSE == decode_mode_is_predecoded) |
0 | 7396 { |
7397 bytes_decoded = Sound_Decode(sample); | |
7398 if(sample->flags & SOUND_SAMPLEFLAG_ERROR) | |
7399 { | |
7400 ALmixer_SetError(Sound_GetError()); | |
7401 Sound_FreeSample(sample); | |
7402 free(ret_data); | |
7403 return NULL; | |
7404 } | |
7405 | |
7406 /* If no data, return an error */ | |
7407 if(0 == bytes_decoded) | |
7408 { | |
7409 ALmixer_SetError("File has no data"); | |
7410 Sound_FreeSample(sample); | |
7411 free(ret_data); | |
7412 return NULL; | |
7413 } | |
7414 | |
7415 /* Note, currently, my Ogg conservative modifications | |
7416 * prevent EOF from being detected in the first read | |
7417 * because of the weird packet behavior of ov_read(). | |
7418 * The EAGAIN will get set, but not the EOF. | |
7419 * I don't know the best way to handle this, | |
7420 * so for now, Ogg's can only be explicitly | |
7421 * predecoded. | |
7422 */ | |
7423 | |
7424 /* Correction: Since we no longer actually keep the | |
7425 * streamed data we read here (we rewind and throw | |
7426 * it away, and start over on Play), it is | |
7427 * safe to read another chunk to see if we've hit EOF | |
7428 */ | |
7429 if(sample->flags & SOUND_SAMPLEFLAG_EAGAIN) | |
7430 { | |
7431 bytes_decoded = Sound_Decode(sample); | |
7432 if(sample->flags & SOUND_SAMPLEFLAG_ERROR) | |
7433 { | |
7434 ALmixer_SetError(Sound_GetError()); | |
7435 Sound_FreeSample(sample); | |
7436 free(ret_data); | |
7437 return NULL; | |
7438 } | |
7439 } | |
7440 | |
7441 | |
7442 /* If we found an EOF, the entire file was | |
7443 * decoded, so we can treat it like one. | |
7444 */ | |
7445 | |
7446 if(sample->flags & SOUND_SAMPLEFLAG_EOF) | |
7447 { | |
7448 fprintf(stderr, "We got LUCKY! File is predecoded even though STREAM was requested\n"); | |
7449 | |
7450 ret_data->decoded_all = 1; | |
7451 /* Need to keep this information around for | |
7452 * seek and rewind abilities. | |
7453 */ | |
7454 ret_data->total_bytes = bytes_decoded; | |
7455 /* For now, the loaded bytes is the same as total bytes, but | |
7456 * this could change during a seek operation | |
7457 */ | |
7458 ret_data->loaded_bytes = bytes_decoded; | |
7459 | |
7460 /* Let's compute the total playing time | |
7461 * SDL_sound does not yet provide this (we're working on | |
7462 * that at the moment...) | |
7463 */ | |
7464 ret_data->total_time = Compute_Total_Time(&sample->desired, bytes_decoded); | |
7465 | |
7466 /* Create one element in the buffer array for data for OpanAL */ | |
7467 ret_data->buffer = (ALuint*)malloc( sizeof(ALuint) ); | |
7468 if(NULL == ret_data->buffer) | |
7469 { | |
7470 ALmixer_SetError("Out of Memory"); | |
7471 Sound_FreeSample(sample); | |
7472 free(ret_data); | |
7473 return NULL; | |
7474 } | |
7475 /* Clear the error code */ | |
7476 alGetError(); | |
7477 /* Now generate an OpenAL buffer using that first element */ | |
7478 alGenBuffers(1, ret_data->buffer); | |
7479 if( (error = alGetError()) != AL_NO_ERROR) | |
7480 { | |
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7481 ALmixer_SetError("alGenBuffers failed: %s\n", alGetString(error)); |
0 | 7482 Sound_FreeSample(sample); |
7483 free(ret_data->buffer); | |
7484 free(ret_data); | |
7485 return NULL; | |
7486 } | |
7487 | |
7488 | |
7489 /* Now copy the data to the OpenAL buffer */ | |
7490 /* We can't just set a pointer because the API needs | |
7491 * its own copy to assist hardware acceleration */ | |
7492 alBufferData(ret_data->buffer[0], | |
7493 TranslateFormat(&sample->desired), | |
7494 sample->buffer, | |
7495 bytes_decoded, | |
7496 sample->desired.rate | |
7497 ); | |
7498 if( (error = alGetError()) != AL_NO_ERROR) | |
7499 { | |
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|
7500 ALmixer_SetError("alBufferData failed: %s\n", alGetString(error)); |
0 | 7501 Sound_FreeSample(sample); |
7502 alDeleteBuffers(1, ret_data->buffer); | |
7503 free(ret_data->buffer); | |
7504 free(ret_data); | |
7505 return NULL; | |
7506 } | |
7507 | |
7508 /* We should be done with the sample since it's all | |
7509 * predecoded. So we can free the memory */ | |
7510 | |
7511 /* Additional notes: | |
7512 * We need to keep data around in case Seek() is needed | |
7513 * or other Sound_AudioInfo is needed. | |
7514 * This can either be done by not deleting the sample, | |
7515 * or it can be done by dynamically recreating it | |
7516 * when we need it. | |
7517 */ | |
7518 /* Since OpenAL won't let us retrieve it | |
7519 * (aka dynamically), we have to keep the Sample | |
7520 * around because since the user requested | |
7521 * streamed and we offered predecoded, | |
7522 * we don't want to mess up the user who | |
7523 * was expecting seek support | |
7524 * So Don't Do anything | |
7525 */ | |
7526 /* | |
7527 if(0 == access_data) | |
7528 { | |
7529 Sound_FreeSample(sample); | |
7530 ret_data->sample = NULL; | |
7531 } | |
7532 */ | |
7533 /* Else, We keep a copy of the sample around. | |
7534 * so don't do anything. | |
7535 */ | |
7536 | |
7537 #if 0 | |
7538 #if defined(DISABLE_PREDECODED_SEEK) | |
7539 Sound_FreeSample(sample); | |
7540 ret_data->sample = NULL; | |
7541 #elif !defined(DISABLE_SEEK_MEMORY_OPTIMIZATION) | |
7542 Sound_FreeSample(sample); | |
7543 ret_data->sample = NULL; | |
7544 #else | |
7545 /* We keep a copy of the sample around. | |
7546 * so don't do anything. | |
7547 */ | |
7548 #endif | |
7549 #endif | |
7550 /* okay we're done here */ | |
7551 | |
7552 } | |
7553 /* Else, we need to stream the data, so we'll | |
7554 * create multple buffers for queuing */ | |
7555 else | |
7556 { | |
7557 fprintf(stderr, "Loading streamed data (not lucky)\n"); | |
7558 ret_data->decoded_all = 0; | |
7559 | |
7560 /* This information is for predecoded. | |
7561 * Set to 0, since we don't know. | |
7562 */ | |
7563 ret_data->total_bytes = 0; | |
7564 | |
7565 /* Create buffers for data | |
7566 */ | |
7567 ret_data->buffer = (ALuint*)malloc( sizeof(ALuint) * max_queue_buffers); | |
7568 if(NULL == ret_data->buffer) | |
7569 { | |
7570 ALmixer_SetError("Out of Memory"); | |
7571 Sound_FreeSample(sample); | |
7572 free(ret_data); | |
7573 return NULL; | |
7574 } | |
7575 | |
7576 /* Clear the error code */ | |
7577 alGetError(); | |
7578 /* Now generate an OpenAL buffer using that first element */ | |
7579 alGenBuffers(max_queue_buffers, ret_data->buffer); | |
7580 if( (error = alGetError()) != AL_NO_ERROR) | |
7581 { | |
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7582 ALmixer_SetError("alGenBuffers failed: %s\n", alGetString(error)); |
0 | 7583 Sound_FreeSample(sample); |
7584 free(ret_data->buffer); | |
7585 free(ret_data); | |
7586 return NULL; | |
7587 } | |
7588 | |
7589 /* Redesign: Okay, because of the unqueuing problems and such, | |
7590 * I've decided to redesign where and how queuing is handled. | |
7591 * Before, everything was queued up here. However, this | |
7592 * placed a penalty on load and made performance inconsistent | |
7593 * when samples had to be rewound. It did make things easier | |
7594 * to queue because I could let OpenAL decide which buffer | |
7595 * needed to be queued next. | |
7596 * Now, I'm going to push off the queuing to the actual | |
7597 * Play() command. I'm going to add some book keeping, | |
7598 * and allow for additional buffers to be filled at later | |
7599 * times. | |
7600 */ | |
7601 | |
7602 | |
7603 /* So first of all, because of I already decoded the sample | |
7604 * for testing, I need to decide what to do with it. | |
7605 * The best thing would be be to alBufferData() it. | |
7606 * The problem is it may conflict with the rest of | |
7607 * the system because everything now assumes buffers | |
7608 * are entirely stripped (because of the unqueing | |
7609 * problem). | |
7610 * So it looks like I have to do the crappy thing | |
7611 * and throw away the data, and rewind. | |
7612 */ | |
7613 | |
7614 if(0 == Sound_Rewind(ret_data->sample)) | |
7615 { | |
7616 ALmixer_SetError("Cannot use sample for streamed data because it must be rewindable: %s", Sound_GetError() ); | |
7617 Sound_FreeSample(sample); | |
7618 free(ret_data->buffer); | |
7619 free(ret_data); | |
7620 return NULL; | |
7621 } | |
7622 | |
7623 | |
7624 /* If the user has selected access_data, we need to | |
7625 * keep copies of the queuing buffers around because | |
7626 * OpenAL won't let us access the data. | |
7627 * Allocate the memory for the buffers here | |
7628 * and initialize the albuffer-index map | |
7629 */ | |
7630 if(access_data) | |
7631 { | |
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|
7632 ALuint j; |
0 | 7633 /* Create buffers for data access |
7634 * Should be the same number as the number of queue buffers | |
7635 */ | |
1 | 7636 ret_data->buffer_map_list = (ALmixer_Buffer_Map*)malloc( sizeof(ALmixer_Buffer_Map) * max_queue_buffers); |
0 | 7637 if(NULL == ret_data->buffer_map_list) |
7638 { | |
7639 ALmixer_SetError("Out of Memory"); | |
7640 Sound_FreeSample(sample); | |
7641 free(ret_data->buffer); | |
7642 free(ret_data); | |
7643 return NULL; | |
7644 } | |
7645 | |
7646 ret_data->circular_buffer_queue = CircularQueueUnsignedInt_CreateQueue(max_queue_buffers); | |
7647 if(NULL == ret_data->circular_buffer_queue) | |
7648 { | |
7649 ALmixer_SetError("Out of Memory"); | |
7650 free(ret_data->buffer_map_list); | |
7651 Sound_FreeSample(sample); | |
7652 free(ret_data->buffer); | |
7653 free(ret_data); | |
7654 return NULL; | |
7655 } | |
7656 | |
7657 | |
7658 for(j=0; j<max_queue_buffers; j++) | |
7659 { | |
7660 ret_data->buffer_map_list[j].albuffer = ret_data->buffer[j]; | |
7661 ret_data->buffer_map_list[j].index = j; | |
7662 ret_data->buffer_map_list[j].num_bytes = 0; | |
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7663 ret_data->buffer_map_list[j].data = (ALbyte*)malloc( sizeof(ALbyte) * buffersize); |
0 | 7664 if(NULL == ret_data->buffer_map_list[j].data) |
7665 { | |
7666 ALmixer_SetError("Out of Memory"); | |
7667 break; | |
7668 } | |
7669 } | |
7670 /* If an error happened, we have to clean up the memory */ | |
7671 if(j < max_queue_buffers) | |
7672 { | |
7673 fprintf(stderr, "################## Buffer allocation failed\n"); | |
7674 for( ; j>=0; j--) | |
7675 { | |
7676 free(ret_data->buffer_map_list[j].data); | |
7677 } | |
7678 free(ret_data->buffer_map_list); | |
7679 CircularQueueUnsignedInt_FreeQueue(ret_data->circular_buffer_queue); | |
7680 Sound_FreeSample(sample); | |
7681 free(ret_data->buffer); | |
7682 free(ret_data); | |
7683 return NULL; | |
7684 } | |
7685 | |
7686 /* The Buffer_Map_List must be sorted by albuffer for binary searches | |
7687 */ | |
1 | 7688 qsort(ret_data->buffer_map_list, max_queue_buffers, sizeof(ALmixer_Buffer_Map), Compare_Buffer_Map); |
0 | 7689 } /* End if access_data==true */ |
7690 | |
7691 | |
7692 } /* End of do stream */ | |
7693 } /* end of DECODE_STREAM */ | |
7694 /* User requested decode all (easy, nothing to figure out) */ | |
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7695 else if(AL_TRUE == decode_mode_is_predecoded) |
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7696 { |
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7697 #ifndef ALMIXER_DISABLE_PREDECODED_PRECOMPUTE_BUFFER_SIZE_OPTIMIZATION |
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7698 /* SDL_sound (behind the scenes) seems to loop on buffer_size chunks |
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7699 * until the buffer is filled. It seems like we can |
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7700 * do much better and precompute the size of the buffer |
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7701 * so looping isn't needed. |
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|
7702 * WARNING: Due to the way SDL_sound is currently implemented, |
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7703 * this may waste a lot of memory up front. |
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7704 * SDL_sound seems to pre-create a buffer of the requested size, |
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7705 * but on DecodeAll, an entirely new buffer is created and |
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7706 * everything is memcpy'd into the new buffer in read chunks |
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|
7707 * of the buffer_size. This means we need roughly twice the memory |
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|
7708 * to load a file. |
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|
7709 */ |
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|
7710 ALint sound_duration = Sound_GetDuration(sample); |
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7711 if(sound_duration > 0) |
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|
7712 { |
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|
7713 size_t total_bytes = Compute_Total_Bytes_With_Frame_Padding(&sample->desired, (ALuint)sound_duration); |
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7714 int buffer_resize_succeeded = Sound_SetBufferSize(sample, total_bytes); |
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|
7715 if(0 == buffer_resize_succeeded) |
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|
7716 { |
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|
7717 ALmixer_SetError(Sound_GetError()); |
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|
7718 Sound_FreeSample(sample); |
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|
7719 free(ret_data); |
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7720 return NULL; |
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|
7721 } |
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|
7722 } |
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|
7723 #endif /* ALMIXER_DISABLE_PREDECODED_PRECOMPUTE_BUFFER_SIZE_OPTIMIZATION */ |
0 | 7724 bytes_decoded = Sound_DecodeAll(sample); |
7725 if(sample->flags & SOUND_SAMPLEFLAG_ERROR) | |
7726 { | |
7727 ALmixer_SetError(Sound_GetError()); | |
7728 Sound_FreeSample(sample); | |
7729 free(ret_data); | |
7730 return NULL; | |
7731 } | |
7732 | |
7733 /* If no data, return an error */ | |
7734 if(0 == bytes_decoded) | |
7735 { | |
7736 ALmixer_SetError("File has no data"); | |
7737 Sound_FreeSample(sample); | |
7738 free(ret_data); | |
7739 return NULL; | |
7740 } | |
7741 | |
7742 | |
7743 ret_data->decoded_all = 1; | |
7744 /* Need to keep this information around for | |
7745 * seek and rewind abilities. | |
7746 */ | |
7747 ret_data->total_bytes = bytes_decoded; | |
7748 /* For now, the loaded bytes is the same as total bytes, but | |
7749 * this could change during a seek operation | |
7750 */ | |
7751 ret_data->loaded_bytes = bytes_decoded; | |
7752 | |
7753 /* Let's compute the total playing time | |
7754 * SDL_sound does not yet provide this (we're working on | |
7755 * that at the moment...) | |
7756 */ | |
7757 ret_data->total_time = Compute_Total_Time(&sample->desired, bytes_decoded); | |
7758 | |
7759 /* Create one element in the buffer array for data for OpanAL */ | |
7760 ret_data->buffer = (ALuint*)malloc( sizeof(ALuint) ); | |
7761 if(NULL == ret_data->buffer) | |
7762 { | |
7763 ALmixer_SetError("Out of Memory"); | |
7764 Sound_FreeSample(sample); | |
7765 free(ret_data); | |
7766 return NULL; | |
7767 } | |
7768 /* Clear the error code */ | |
7769 alGetError(); | |
7770 /* Now generate an OpenAL buffer using that first element */ | |
7771 alGenBuffers(1, ret_data->buffer); | |
7772 if( (error = alGetError()) != AL_NO_ERROR) | |
7773 { | |
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7774 ALmixer_SetError("alGenBuffers failed: %s\n", alGetString(error)); |
0 | 7775 Sound_FreeSample(sample); |
7776 free(ret_data->buffer); | |
7777 free(ret_data); | |
7778 return NULL; | |
7779 } | |
7780 fprintf(stderr, "Actual rate=%d, desired=%d\n", sample->actual.rate, sample->desired.rate); | |
7781 | |
7782 /* Now copy the data to the OpenAL buffer */ | |
7783 /* We can't just set a pointer because the API needs | |
7784 * its own copy to assist hardware acceleration */ | |
7785 alBufferData(ret_data->buffer[0], | |
7786 TranslateFormat(&sample->desired), | |
7787 sample->buffer, | |
7788 bytes_decoded, | |
7789 sample->desired.rate | |
7790 ); | |
7791 if( (error = alGetError()) != AL_NO_ERROR) | |
7792 { | |
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7793 ALmixer_SetError("alBufferData failed: %s\n", alGetString(error)); |
0 | 7794 Sound_FreeSample(sample); |
7795 alDeleteBuffers(1, ret_data->buffer); | |
7796 free(ret_data->buffer); | |
7797 free(ret_data); | |
7798 return NULL; | |
7799 } | |
7800 | |
7801 /* We should be done with the sample since it's all | |
7802 * predecoded. So we can free the memory */ | |
7803 /* Need to keep around because Seek() needs it */ | |
7804 | |
7805 /* Additional notes: | |
7806 * We need to keep data around in case Seek() is needed | |
7807 * or other Sound_AudioInfo is needed. | |
7808 * This can either be done by not deleting the sample, | |
7809 * or it can be done by dynamically recreating it | |
7810 * when we need it. | |
7811 * Update: I think now it's up to the user by passing the | |
7812 * access_data flag. If they set the flag, then they get | |
7813 * data callbacks and seek support. If not, then they can | |
7814 * get all that stuff at the expense of keeping extra memory | |
7815 * around. | |
7816 */ | |
7817 if(0 == access_data) | |
7818 { | |
7819 Sound_FreeSample(sample); | |
7820 ret_data->sample = NULL; | |
7821 } | |
7822 | |
7823 /* Else, We keep a copy of the sample around. | |
7824 * so don't do anything. | |
7825 */ | |
7826 #if 0 | |
7827 #if defined(DISABLE_PREDECODED_SEEK) | |
7828 Sound_FreeSample(sample); | |
7829 ret_data->sample = NULL; | |
7830 #elif !defined(DISABLE_SEEK_MEMORY_OPTIMIZATION) | |
7831 Sound_FreeSample(sample); | |
7832 ret_data->sample = NULL; | |
7833 #else | |
7834 /* We keep a copy of the sample around. | |
7835 * so don't do anything. | |
7836 */ | |
7837 #endif | |
7838 #endif | |
7839 | |
7840 fprintf(stderr, "Made it\n"); | |
7841 /* okay we're done here */ | |
7842 } | |
7843 else | |
7844 { | |
7845 /* Shouldn't get here */ | |
7846 ALmixer_SetError("Unknown decode mode"); | |
7847 Sound_FreeSample(sample); | |
7848 free(ret_data); | |
7849 return NULL; | |
7850 } | |
7851 | |
7852 fprintf(stderr, "Returning data\n"); | |
7853 return ret_data; | |
7854 } | |
7855 | |
7856 | |
7857 /* This will load a sample for us. Most of the uglyness is | |
7858 * error checking and the fact that streamed/predecoded files | |
7859 * must be treated differently. | |
7860 * I don't like the AudioInfo parameter. I removed it once, | |
7861 * but the system will fail on RAW samples because the user | |
7862 * must specify it, so I had to bring it back. | |
7863 * Remember I must close the rwops if there is an error before NewSample() | |
7864 */ | |
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7865 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 | 7866 { |
7867 Sound_Sample* sample = NULL; | |
7868 Sound_AudioInfo target; | |
7869 | |
7870 /* Initialize target values to defaults | |
7871 * 0 tells SDL_sound to use the "actual" values | |
7872 */ | |
7873 target.channels = 0; | |
7874 target.rate = 0; | |
7875 #if 0 | |
7876 /* This requires my new additions to SDL_sound. It will | |
7877 * convert the sample to the proper endian order. | |
7878 * If the actual is 8-bit, it will do unsigned, if | |
7879 * the actual is 16-bit, it will do signed. | |
7880 * I'm told by Ryan Gordon that OpenAL prefers the signedness | |
7881 * in this way. | |
7882 */ | |
7883 target.format = AUDIO_U8S16SYS; | |
7884 #else | |
7885 target.format = AUDIO_S16SYS; | |
7886 #endif | |
7887 | |
7888 /* Set a default buffersize if needed */ | |
7889 if(0 == buffersize) | |
7890 { | |
7891 buffersize = ALMIXER_DEFAULT_BUFFERSIZE; | |
7892 } | |
7893 | |
7894 sample = Sound_NewSample(rwops, fileext, &target, buffersize); | |
7895 if(NULL == sample) | |
7896 { | |
7897 ALmixer_SetError(Sound_GetError()); | |
7898 return NULL; | |
7899 } | |
7900 | |
1 | 7901 return( DoLoad(sample, buffersize, decode_mode_is_predecoded, max_queue_buffers, num_startup_buffers, access_data)); |
0 | 7902 } |
7903 | |
7904 | |
7905 | |
7906 /* This will load a sample for us from | |
7907 * a file (instead of RWops). Most of the uglyness is | |
7908 * error checking and the fact that streamed/predecoded files | |
7909 * must be treated differently. | |
7910 */ | |
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7911 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 | 7912 { |
7913 Sound_Sample* sample = NULL; | |
7914 Sound_AudioInfo target; | |
7915 | |
7916 /* Initialize target values to defaults | |
7917 * 0 tells SDL_sound to use the "actual" values | |
7918 */ | |
7919 target.channels = 0; | |
7920 target.rate = 0; | |
7921 | |
7922 #if 0 | |
7923 /* This requires my new additions to SDL_sound. It will | |
7924 * convert the sample to the proper endian order. | |
7925 * If the actual is 8-bit, it will do unsigned, if | |
7926 * the actual is 16-bit, it will do signed. | |
7927 * I'm told by Ryan Gordon that OpenAL prefers the signedness | |
7928 * in this way. | |
7929 */ | |
7930 target.format = AUDIO_U8S16SYS; | |
7931 #else | |
7932 target.format = AUDIO_S16SYS; | |
7933 #endif | |
7934 | |
7935 #if 0 | |
7936 /* Okay, here's a messy hack. The problem is that we need | |
7937 * to convert the sample to have the correct bitdepth, | |
7938 * endian order, and signedness values. | |
7939 * The bit depth is 8 or 16. | |
7940 * The endian order is the native order of the system. | |
7941 * The signedness depends on what the original value | |
7942 * of the sample. Unfortunately, we can't specify these | |
7943 * values until we after we already know what the original | |
7944 * values were for bitdepth and signedness. | |
7945 * So we must open the file once to get the values, | |
7946 * then close it, and then reopen it with the | |
7947 * correct desired target values. | |
7948 * I tried changing the sample->desired field after | |
7949 * the NewSample call, but it had no effect, so | |
7950 * it looks like it must be set on open. | |
7951 */ | |
7952 /* Pick a small buffersize for the first open to not | |
7953 * waste much time allocating memory */ | |
7954 sample = Sound_NewSampleFromFile(filename, NULL, 512); | |
7955 if(NULL == sample) | |
7956 { | |
7957 ALmixer_SetError(Sound_GetError()); | |
7958 return NULL; | |
7959 } | |
7960 | |
7961 bit_depth = GetBitDepth(sample->actual.format); | |
7962 signedness_value = GetSignednessValue(sample->actual.format); | |
7963 if(8 == bit_depth) | |
7964 { | |
7965 /* If 8 bit, then we don't have to worry about | |
7966 * endian issues. We can just use the actual format | |
7967 * value and it should do the right thing | |
7968 */ | |
7969 target.format = sample->actual.format; | |
7970 } | |
7971 else | |
7972 { | |
7973 /* We'll assume it's 16-bit, and if it's not | |
7974 * hopefully SDL_sound will return an error, | |
7975 * or let us convert to 16-bit | |
7976 */ | |
7977 /* Now we need to get the correct signedness */ | |
7978 if(ALMIXER_UNSIGNED_VALUE == signedness_value) | |
7979 { | |
7980 /* Set to Unsigned 16-bit, system endian order */ | |
7981 target.format = AUDIO_U16SYS; | |
7982 } | |
7983 else | |
7984 { | |
7985 /* Again, we'll assume it's Signed 16-bit system order | |
7986 * or force the conversion and hope it works out | |
7987 */ | |
7988 target.format = AUDIO_S16SYS; | |
7989 } | |
7990 } | |
7991 | |
7992 /* Now we have the correct info. We need to close and reopen */ | |
7993 Sound_FreeSample(sample); | |
7994 #endif | |
7995 | |
7996 sample = Sound_NewSampleFromFile(filename, &target, buffersize); | |
7997 if(NULL == sample) | |
7998 { | |
7999 ALmixer_SetError(Sound_GetError()); | |
8000 return NULL; | |
8001 } | |
8002 | |
8003 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); | |
8004 | |
1 | 8005 return( DoLoad(sample, buffersize, decode_mode_is_predecoded, max_queue_buffers, num_startup_buffers, access_data)); |
0 | 8006 } |
8007 | |
8008 | |
8009 /* This is a back door for RAW samples or if you need the | |
8010 * AudioInfo field. Use at your own risk. | |
8011 */ | |
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8012 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 | 8013 { |
8014 Sound_Sample* sample = NULL; | |
1 | 8015 Sound_AudioInfo sound_desired; |
8016 /* Rather than copying the data from struct to struct, I could just | |
8017 * cast the thing since the structs are meant to be identical. | |
8018 * But if SDL_sound changes it's implementation, bad things | |
8019 * will probably happen. (Or if I change my implementation and | |
8020 * forget about the cast, same bad scenario.) Since this is a load | |
8021 * function, performance of this is negligible. | |
8022 */ | |
8023 if(NULL == desired) | |
8024 { | |
8025 sample = Sound_NewSample(rwops, fileext, NULL, buffersize); | |
8026 } | |
8027 else | |
8028 { | |
8029 sound_desired.format = desired->format; | |
8030 sound_desired.channels = desired->channels; | |
8031 sound_desired.rate = desired->rate; | |
8032 sample = Sound_NewSample(rwops, fileext, &sound_desired, buffersize); | |
8033 } | |
0 | 8034 if(NULL == sample) |
8035 { | |
8036 ALmixer_SetError(Sound_GetError()); | |
8037 return NULL; | |
8038 } | |
1 | 8039 return( DoLoad(sample, buffersize, decode_mode_is_predecoded, max_queue_buffers, num_startup_buffers, access_data)); |
0 | 8040 } |
8041 | |
8042 | |
8043 | |
8044 | |
8045 /* This is a back door for RAW samples or if you need the | |
8046 * AudioInfo field. Use at your own risk. | |
8047 */ | |
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8048 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 | 8049 { |
8050 Sound_Sample* sample = NULL; | |
1 | 8051 Sound_AudioInfo sound_desired; |
8052 /* Rather than copying the data from struct to struct, I could just | |
8053 * cast the thing since the structs are meant to be identical. | |
8054 * But if SDL_sound changes it's implementation, bad things | |
8055 * will probably happen. (Or if I change my implementation and | |
8056 * forget about the cast, same bad scenario.) Since this is a load | |
8057 * function, performance of this is negligible. | |
8058 */ | |
8059 if(NULL == desired) | |
8060 { | |
8061 sample = Sound_NewSampleFromFile(filename, NULL, buffersize); | |
8062 } | |
8063 else | |
8064 { | |
8065 sound_desired.format = desired->format; | |
8066 sound_desired.channels = desired->channels; | |
8067 sound_desired.rate = desired->rate; | |
8068 sample = Sound_NewSampleFromFile(filename, &sound_desired, buffersize); | |
8069 } | |
8070 | |
0 | 8071 if(NULL == sample) |
8072 { | |
8073 ALmixer_SetError(Sound_GetError()); | |
8074 return NULL; | |
8075 } | |
1 | 8076 return( DoLoad(sample, buffersize, decode_mode_is_predecoded, max_queue_buffers, num_startup_buffers, access_data)); |
0 | 8077 } |
8078 | |
8079 | |
8080 | |
8081 | |
8082 void ALmixer_FreeData(ALmixer_Data* data) | |
8083 { | |
8084 ALenum error; | |
8085 if(NULL == data) | |
8086 { | |
8087 return; | |
8088 } | |
8089 | |
8090 if(data->decoded_all) | |
8091 { | |
8092 /* If access_data was enabled, then the Sound_Sample* | |
8093 * still exists. We need to free it | |
8094 */ | |
8095 if(data->sample != NULL) | |
8096 { | |
8097 Sound_FreeSample(data->sample); | |
8098 } | |
8099 alDeleteBuffers(1, data->buffer); | |
8100 if((error = alGetError()) != AL_NO_ERROR) | |
8101 { | |
8102 fprintf(stderr, "70Testing error: %s\n", | |
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8103 alGetString(error)); |
0 | 8104 } |
8105 | |
8106 } | |
8107 else | |
8108 { | |
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8109 ALuint i; |
0 | 8110 |
8111 /* Delete buffer copies if access_data was enabled */ | |
8112 if(data->buffer_map_list != NULL) | |
8113 { | |
8114 for(i=0; i<data->max_queue_buffers; i++) | |
8115 { | |
8116 free(data->buffer_map_list[i].data); | |
8117 } | |
8118 free(data->buffer_map_list); | |
8119 } | |
8120 if(data->circular_buffer_queue != NULL) | |
8121 { | |
8122 CircularQueueUnsignedInt_FreeQueue(data->circular_buffer_queue); | |
8123 } | |
8124 | |
8125 Sound_FreeSample(data->sample); | |
8126 alDeleteBuffers(data->max_queue_buffers, data->buffer); | |
8127 if((error = alGetError()) != AL_NO_ERROR) | |
8128 { | |
8129 fprintf(stderr, "71Testing error: %s\n", | |
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8130 alGetString(error)); |
0 | 8131 } |
8132 } | |
8133 free(data->buffer); | |
8134 free(data); | |
8135 } | |
8136 | |
2
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8137 ALint ALmixer_GetTotalTime(ALmixer_Data* data) |
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8138 { |
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8139 if(NULL == data) |
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8140 { |
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8141 return -1; |
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8142 } |
0 | 8143 return data->total_time; |
8144 } | |
8145 | |
8146 /* This function will look up the source for the corresponding channel */ | |
8147 /* Must return 0 on error instead of -1 because of unsigned int */ | |
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8148 ALuint ALmixer_GetSource(ALint channel) |
0 | 8149 { |
8150 ALuint retval; | |
2
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8151 #ifdef ENABLE_ALMIXER_THREADS |
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8152 SDL_LockMutex(s_simpleLock); |
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8153 #endif |
0 | 8154 retval = Internal_GetSource(channel); |
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8155 #ifdef ENABLE_ALMIXER_THREADS |
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8156 SDL_UnlockMutex(s_simpleLock); |
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8157 #endif |
0 | 8158 return retval; |
8159 } | |
8160 | |
8161 /* This function will look up the channel for the corresponding source */ | |
2
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8162 ALint ALmixer_GetChannel(ALuint source) |
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8163 { |
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8164 ALint retval; |
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8165 #ifdef ENABLE_ALMIXER_THREADS |
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8166 SDL_LockMutex(s_simpleLock); |
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8167 #endif |
0 | 8168 retval = Internal_GetChannel(source); |
2
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8169 #ifdef ENABLE_ALMIXER_THREADS |
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8170 SDL_UnlockMutex(s_simpleLock); |
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8171 #endif |
0 | 8172 return retval; |
8173 } | |
8174 | |
2
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8175 ALint ALmixer_FindFreeChannel(ALint start_channel) |
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8176 { |
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8177 ALint retval; |
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8178 #ifdef ENABLE_ALMIXER_THREADS |
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8179 SDL_LockMutex(s_simpleLock); |
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8180 #endif |
0 | 8181 retval = Internal_FindFreeChannel(start_channel); |
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8182 #ifdef ENABLE_ALMIXER_THREADS |
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8183 SDL_UnlockMutex(s_simpleLock); |
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8184 #endif |
0 | 8185 return retval; |
8186 } | |
8187 | |
8188 | |
8189 | |
8190 /* API update function. | |
8191 * It should return the number of buffers that were | |
8192 * queued during the call. The value might be | |
8193 * used to guage how long you might wait to | |
8194 * call the next update loop in case you are worried | |
8195 * about preserving CPU cycles. The idea is that | |
8196 * when a buffer is queued, there was probably some | |
8197 * CPU intensive looping which took awhile. | |
8198 * It's mainly provided as a convenience. | |
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8199 * Timing the call with ALmixer_GetTicks() would produce |
0 | 8200 * more accurate information. |
8201 * Returns a negative value if there was an error, | |
8202 * the value being the number of errors. | |
8203 */ | |
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8204 ALint ALmixer_Update() |
0 | 8205 { |
8206 #ifdef ENABLE_ALMIXER_THREADS | |
8207 /* The thread will handle all updates by itself. | |
8208 * Don't allow the user to explicitly call update. | |
8209 */ | |
8210 return 0; | |
8211 #else | |
8212 return( Update_ALmixer(NULL) ); | |
8213 #endif | |
8214 } | |
8215 | |
8216 | |
8217 | |
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8218 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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8219 { |
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8220 #ifdef ENABLE_ALMIXER_THREADS |
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8221 SDL_LockMutex(s_simpleLock); |
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8222 #endif |
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8223 Channel_Done_Callback = playback_finished_callback; |
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8224 Channel_Done_Callback_Userdata = user_data; |
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8225 #ifdef ENABLE_ALMIXER_THREADS |
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8226 SDL_UnlockMutex(s_simpleLock); |
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8227 #endif |
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8228 } |
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8229 |
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8230 |
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8231 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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8232 { |
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8233 #ifdef ENABLE_ALMIXER_THREADS |
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|
8234 SDL_LockMutex(s_simpleLock); |
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Eric Wing <ewing . public |-at-| gmail . com>
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1
diff
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|
8235 #endif |
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1
diff
changeset
|
8236 Channel_Data_Callback = playback_data_callback; |
1 | 8237 Channel_Data_Callback_Userdata = user_data; |
2
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1
diff
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|
8238 #ifdef ENABLE_ALMIXER_THREADS |
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diff
changeset
|
8239 SDL_UnlockMutex(s_simpleLock); |
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parents:
1
diff
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|
8240 #endif |
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8241 } |
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Eric Wing <ewing . public |-at-| gmail . com>
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1
diff
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|
8242 |
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1
diff
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|
8243 |
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
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1
diff
changeset
|
8244 |
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1
diff
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|
8245 |
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Eric Wing <ewing . public |-at-| gmail . com>
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diff
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|
8246 |
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
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1
diff
changeset
|
8247 ALint ALmixer_PlayChannelTimed(ALint channel, ALmixer_Data* data, ALint loops, ALint ticks) |
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Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
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1
diff
changeset
|
8248 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8249 ALint retval; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8250 #ifdef ENABLE_ALMIXER_THREADS |
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1
diff
changeset
|
8251 SDL_LockMutex(s_simpleLock); |
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1
diff
changeset
|
8252 #endif |
0 | 8253 retval = Internal_PlayChannelTimed(channel, data, loops, ticks); |
2
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1
diff
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|
8254 #ifdef ENABLE_ALMIXER_THREADS |
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1
diff
changeset
|
8255 SDL_UnlockMutex(s_simpleLock); |
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1
diff
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|
8256 #endif |
0 | 8257 return retval; |
8258 } | |
8259 | |
8260 | |
8261 /* In case the user wants to specify a source instead of a channel, | |
8262 * they may use this function. This function will look up the | |
8263 * source-to-channel map, and convert the call into a | |
8264 * PlayChannelTimed() function call. | |
8265 * Returns the channel it's being played on. | |
8266 * Note: If you are prefer this method, then you need to be careful | |
8267 * about using PlayChannel, particularly if you request the | |
8268 * first available channels because source and channels have | |
8269 * a one-to-one mapping in this API. It is quite easy for | |
8270 * a channel/source to already be in use because of this. | |
8271 * In this event, an error message will be returned to you. | |
8272 */ | |
2
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Eric Wing <ewing . public |-at-| gmail . com>
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1
diff
changeset
|
8273 ALuint ALmixer_PlaySourceTimed(ALuint source, ALmixer_Data* data, ALint loops, ALint ticks) |
0 | 8274 { |
8275 ALuint retval; | |
2
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Eric Wing <ewing . public |-at-| gmail . com>
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1
diff
changeset
|
8276 #ifdef ENABLE_ALMIXER_THREADS |
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1
diff
changeset
|
8277 SDL_LockMutex(s_simpleLock); |
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Eric Wing <ewing . public |-at-| gmail . com>
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1
diff
changeset
|
8278 #endif |
0 | 8279 retval = Internal_PlaySourceTimed(source, data, loops, ticks); |
2
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Eric Wing <ewing . public |-at-| gmail . com>
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1
diff
changeset
|
8280 #ifdef ENABLE_ALMIXER_THREADS |
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Eric Wing <ewing . public |-at-| gmail . com>
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1
diff
changeset
|
8281 SDL_UnlockMutex(s_simpleLock); |
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Eric Wing <ewing . public |-at-| gmail . com>
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1
diff
changeset
|
8282 #endif |
0 | 8283 return retval; |
8284 } | |
8285 | |
8286 | |
8287 /* Will return the number of channels halted | |
8288 * or 0 for error | |
8289 */ | |
2
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Eric Wing <ewing . public |-at-| gmail . com>
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1
diff
changeset
|
8290 ALint ALmixer_HaltChannel(ALint channel) |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8291 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
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1
diff
changeset
|
8292 ALint retval; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8293 #ifdef ENABLE_ALMIXER_THREADS |
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Eric Wing <ewing . public |-at-| gmail . com>
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1
diff
changeset
|
8294 SDL_LockMutex(s_simpleLock); |
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1
diff
changeset
|
8295 #endif |
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
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1
diff
changeset
|
8296 retval = Internal_HaltChannel(channel, AL_FALSE); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
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1
diff
changeset
|
8297 #ifdef ENABLE_ALMIXER_THREADS |
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Eric Wing <ewing . public |-at-| gmail . com>
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1
diff
changeset
|
8298 SDL_UnlockMutex(s_simpleLock); |
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8299 #endif |
0 | 8300 return retval; |
8301 } | |
8302 | |
8303 /* Will return the number of channels halted | |
8304 * or 0 for error | |
8305 */ | |
2
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Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
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1
diff
changeset
|
8306 ALint ALmixer_HaltSource(ALuint source) |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8307 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
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1
diff
changeset
|
8308 ALint retval; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8309 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8310 SDL_LockMutex(s_simpleLock); |
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8311 #endif |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8312 retval = Internal_HaltSource(source, AL_FALSE); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8313 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8314 SDL_UnlockMutex(s_simpleLock); |
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8315 #endif |
0 | 8316 return retval; |
8317 } | |
8318 | |
8319 | |
8320 /* This will rewind the SDL_Sound sample for streamed | |
8321 * samples and start buffering up the data for the next | |
8322 * playback. This may require samples to be halted | |
8323 */ | |
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8324 ALint ALmixer_RewindData(ALmixer_Data* data) |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8325 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8326 ALint retval; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8327 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8328 SDL_LockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8329 #endif |
0 | 8330 retval = Internal_RewindData(data); |
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8331 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8332 SDL_UnlockMutex(s_simpleLock); |
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8333 #endif |
0 | 8334 return retval; |
8335 } | |
8336 | |
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8337 ALint ALmixer_RewindChannel(ALint channel) |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8338 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8339 ALint retval; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8340 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8341 SDL_LockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8342 #endif |
0 | 8343 retval = Internal_RewindChannel(channel); |
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8344 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8345 SDL_UnlockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8346 #endif |
0 | 8347 return retval; |
8348 } | |
8349 | |
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8350 ALint ALmixer_RewindSource(ALuint source) |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8351 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8352 ALint retval; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8353 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8354 SDL_LockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8355 #endif |
0 | 8356 retval = Internal_RewindSource(source); |
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8357 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8358 SDL_UnlockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8359 #endif |
0 | 8360 return retval; |
8361 } | |
8362 | |
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8363 ALint ALmixer_PauseChannel(ALint channel) |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8364 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8365 ALint retval; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8366 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8367 SDL_LockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8368 #endif |
0 | 8369 retval = Internal_PauseChannel(channel); |
2
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8370 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8371 SDL_UnlockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8372 #endif |
0 | 8373 return retval; |
8374 } | |
8375 | |
2
279d0427ef26
Overhaul prep for first public release.
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1
diff
changeset
|
8376 ALint ALmixer_PauseSource(ALuint source) |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8377 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8378 ALint retval; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8379 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8380 SDL_LockMutex(s_simpleLock); |
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8381 #endif |
0 | 8382 retval = Internal_PauseSource(source); |
2
279d0427ef26
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1
diff
changeset
|
8383 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
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parents:
1
diff
changeset
|
8384 SDL_UnlockMutex(s_simpleLock); |
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1
diff
changeset
|
8385 #endif |
0 | 8386 return retval; |
8387 } | |
8388 | |
2
279d0427ef26
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1
diff
changeset
|
8389 ALint ALmixer_ResumeChannel(ALint channel) |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8390 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8391 ALint retval; |
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8392 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
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1
diff
changeset
|
8393 SDL_LockMutex(s_simpleLock); |
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
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1
diff
changeset
|
8394 #endif |
0 | 8395 retval = Internal_ResumeChannel(channel); |
2
279d0427ef26
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1
diff
changeset
|
8396 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
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1
diff
changeset
|
8397 SDL_UnlockMutex(s_simpleLock); |
279d0427ef26
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1
diff
changeset
|
8398 #endif |
0 | 8399 return retval; |
8400 } | |
8401 | |
2
279d0427ef26
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1
diff
changeset
|
8402 ALint ALmixer_ResumeSource(ALuint source) |
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
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1
diff
changeset
|
8403 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8404 ALint retval; |
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8405 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8406 SDL_LockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8407 #endif |
0 | 8408 retval = Internal_ResumeSource(source); |
2
279d0427ef26
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1
diff
changeset
|
8409 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
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1
diff
changeset
|
8410 SDL_UnlockMutex(s_simpleLock); |
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8411 #endif |
0 | 8412 return retval; |
8413 } | |
8414 | |
8415 /* Might consider setting eof to 0 as a "feature" | |
8416 * This will allow seek to end to stay there because | |
8417 * Play automatically rewinds if at the end */ | |
2
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8418 ALint ALmixer_SeekData(ALmixer_Data* data, ALuint msec) |
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8419 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8420 ALint retval; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8421 #ifdef ENABLE_ALMIXER_THREADS |
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8422 SDL_LockMutex(s_simpleLock); |
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8423 #endif |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8424 retval = Internal_SeekData(data, msec); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8425 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8426 SDL_UnlockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8427 #endif |
0 | 8428 return retval; |
8429 } | |
8430 | |
2
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8431 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
|
8432 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8433 ALint retval; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8434 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8435 SDL_LockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8436 #endif |
0 | 8437 retval = Internal_FadeInChannelTimed(channel, data, loops, fade_ticks, expire_ticks); |
2
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8438 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8439 SDL_UnlockMutex(s_simpleLock); |
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8440 #endif |
0 | 8441 return retval; |
8442 } | |
8443 | |
2
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8444 ALuint ALmixer_FadeInSourceTimed(ALuint source, ALmixer_Data* data, ALint loops, ALuint fade_ticks, ALint expire_ticks) |
0 | 8445 { |
8446 ALuint retval; | |
2
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8447 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8448 SDL_LockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8449 #endif |
0 | 8450 retval = Internal_FadeInSourceTimed(source, data, loops, fade_ticks, expire_ticks); |
2
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8451 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8452 SDL_UnlockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8453 #endif |
0 | 8454 return retval; |
8455 } | |
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8456 |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8457 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
|
8458 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8459 ALint retval; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8460 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8461 SDL_LockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8462 #endif |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8463 retval = Internal_FadeOutChannel(channel, ticks); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8464 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8465 SDL_UnlockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8466 #endif |
0 | 8467 return retval; |
8468 } | |
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8469 |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8470 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
|
8471 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8472 ALint retval; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8473 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8474 SDL_LockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8475 #endif |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8476 retval = Internal_FadeOutSource(source, ticks); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8477 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8478 SDL_UnlockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8479 #endif |
0 | 8480 return retval; |
8481 } | |
8482 | |
2
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8483 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
|
8484 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8485 ALint retval; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8486 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8487 SDL_LockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8488 #endif |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8489 retval = Internal_FadeChannel(channel, ticks, volume); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8490 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8491 SDL_UnlockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8492 #endif |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8493 return retval; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8494 } |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8495 |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8496 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
|
8497 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8498 ALint retval; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8499 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8500 SDL_LockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8501 #endif |
0 | 8502 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
|
8503 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8504 SDL_UnlockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8505 #endif |
0 | 8506 return retval; |
8507 } | |
8508 | |
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8509 |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8510 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
|
8511 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8512 ALboolean retval; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8513 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8514 SDL_LockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8515 #endif |
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8516 retval = Internal_SetVolumeChannel(channel, volume); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8517 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8518 SDL_UnlockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8519 #endif |
0 | 8520 return retval; |
8521 } | |
8522 | |
2
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8523 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
|
8524 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8525 ALboolean retval; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8526 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8527 SDL_LockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8528 #endif |
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8529 retval = Internal_SetVolumeSource(source, volume); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8530 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8531 SDL_UnlockMutex(s_simpleLock); |
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8532 #endif |
0 | 8533 return retval; |
8534 } | |
8535 | |
2
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8536 ALfloat ALmixer_GetVolumeChannel(ALint channel) |
0 | 8537 { |
8538 ALfloat retval; | |
2
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8539 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8540 SDL_LockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8541 #endif |
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8542 retval = Internal_GetVolumeChannel(channel); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8543 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8544 SDL_UnlockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8545 #endif |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8546 return retval; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8547 } |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8548 |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8549 ALfloat ALmixer_GetVolumeSource(ALuint source) |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8550 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8551 ALfloat retval; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8552 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8553 SDL_LockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8554 #endif |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8555 retval = Internal_GetVolumeSource(source); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8556 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8557 SDL_UnlockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8558 #endif |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8559 return retval; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8560 } |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8561 |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8562 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
|
8563 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8564 ALboolean retval; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8565 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8566 SDL_LockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8567 #endif |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8568 retval = Internal_SetMaxVolumeChannel(channel, volume); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8569 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8570 SDL_UnlockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8571 #endif |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8572 return retval; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8573 } |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8574 |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8575 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
|
8576 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8577 ALboolean retval; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8578 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8579 SDL_LockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8580 #endif |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8581 retval = Internal_SetMaxVolumeSource(source, volume); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8582 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8583 SDL_UnlockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8584 #endif |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8585 return retval; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8586 } |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8587 |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8588 ALfloat ALmixer_GetMaxVolumeChannel(ALint channel) |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8589 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8590 ALfloat retval; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8591 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8592 SDL_LockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8593 #endif |
0 | 8594 retval = Internal_GetMaxVolumeChannel(channel); |
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8595 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8596 SDL_UnlockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8597 #endif |
0 | 8598 return retval; |
8599 } | |
8600 | |
8601 ALfloat ALmixer_GetMaxVolumeSource(ALuint source) | |
8602 { | |
8603 ALfloat retval; | |
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8604 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8605 SDL_LockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8606 #endif |
0 | 8607 retval = Internal_GetMaxVolumeSource(source); |
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8608 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8609 SDL_UnlockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8610 #endif |
0 | 8611 return retval; |
8612 } | |
8613 | |
8614 | |
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8615 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
|
8616 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8617 ALboolean retval; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8618 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8619 SDL_LockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8620 #endif |
0 | 8621 retval = Internal_SetMinVolumeChannel(channel, volume); |
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8622 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8623 SDL_UnlockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8624 #endif |
0 | 8625 return retval; |
8626 } | |
8627 | |
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8628 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
|
8629 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8630 ALboolean retval; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8631 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8632 SDL_LockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8633 #endif |
0 | 8634 retval = Internal_SetMinVolumeSource(source, volume); |
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8635 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8636 SDL_UnlockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8637 #endif |
0 | 8638 return retval; |
8639 } | |
8640 | |
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8641 ALfloat ALmixer_GetMinVolumeChannel(ALint channel) |
0 | 8642 { |
8643 ALfloat retval; | |
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8644 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8645 SDL_LockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8646 #endif |
0 | 8647 retval = Internal_GetMinVolumeChannel(channel); |
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8648 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8649 SDL_UnlockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8650 #endif |
0 | 8651 return retval; |
8652 } | |
8653 | |
8654 ALfloat ALmixer_GetMinVolumeSource(ALuint source) | |
8655 { | |
8656 ALfloat retval; | |
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8657 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8658 SDL_LockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8659 #endif |
0 | 8660 retval = Internal_GetMinVolumeSource(source); |
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8661 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8662 SDL_UnlockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8663 #endif |
0 | 8664 return retval; |
8665 } | |
8666 | |
8667 | |
8668 | |
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8669 ALboolean ALmixer_SetMasterVolume(ALfloat volume) |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8670 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8671 ALboolean retval; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8672 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8673 SDL_LockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8674 #endif |
0 | 8675 retval = Internal_SetMasterVolume(volume); |
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8676 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8677 SDL_UnlockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8678 #endif |
0 | 8679 return retval; |
8680 } | |
8681 | |
8682 ALfloat ALmixer_GetMasterVolume() | |
8683 { | |
8684 ALfloat retval; | |
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8685 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8686 SDL_LockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8687 #endif |
0 | 8688 retval = Internal_GetMasterVolume(); |
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8689 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8690 SDL_UnlockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8691 #endif |
0 | 8692 return retval; |
8693 } | |
8694 | |
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8695 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
|
8696 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8697 ALint retval; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8698 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8699 SDL_LockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8700 #endif |
0 | 8701 retval = Internal_ExpireChannel(channel, ticks); |
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8702 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8703 SDL_UnlockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8704 #endif |
0 | 8705 return retval; |
8706 } | |
8707 | |
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8708 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
|
8709 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8710 ALint retval; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8711 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8712 SDL_LockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8713 #endif |
0 | 8714 retval = Internal_ExpireSource(source, ticks); |
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8715 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8716 SDL_UnlockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8717 #endif |
0 | 8718 return retval; |
8719 } | |
8720 | |
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8721 ALint ALmixer_IsActiveChannel(ALint channel) |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8722 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8723 ALint retval; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8724 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8725 SDL_LockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8726 #endif |
0 | 8727 retval = Internal_QueryChannel(channel); |
2
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8728 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8729 SDL_UnlockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8730 #endif |
0 | 8731 return retval; |
8732 } | |
8733 | |
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8734 ALint ALmixer_IsActiveSource(ALuint source) |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8735 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8736 ALint retval; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8737 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8738 SDL_LockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8739 #endif |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8740 retval = Internal_QuerySource(source); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8741 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8742 SDL_UnlockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8743 #endif |
0 | 8744 return retval; |
8745 } | |
8746 | |
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8747 |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8748 ALint ALmixer_IsPlayingChannel(ALint channel) |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8749 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8750 ALint retval; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8751 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8752 SDL_LockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8753 #endif |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8754 retval = Internal_PlayingChannel(channel); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8755 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8756 SDL_UnlockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8757 #endif |
0 | 8758 return retval; |
8759 } | |
8760 | |
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8761 ALint ALmixer_IsPlayingSource(ALuint source) |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8762 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8763 ALint retval; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8764 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8765 SDL_LockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8766 #endif |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8767 retval = Internal_PlayingSource(source); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8768 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8769 SDL_UnlockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8770 #endif |
0 | 8771 return retval; |
8772 } | |
8773 | |
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8774 |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8775 ALint ALmixer_IsPausedChannel(ALint channel) |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8776 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8777 ALint retval; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8778 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8779 SDL_LockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8780 #endif |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8781 retval = Internal_PausedChannel(channel); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8782 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8783 SDL_UnlockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8784 #endif |
0 | 8785 return retval; |
8786 } | |
8787 | |
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8788 ALint ALmixer_IsPausedSource(ALuint source) |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8789 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8790 ALint retval; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8791 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8792 SDL_LockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8793 #endif |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8794 retval = Internal_PausedSource(source); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8795 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8796 SDL_UnlockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8797 #endif |
0 | 8798 return retval; |
8799 } | |
8800 | |
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8801 |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8802 ALuint ALmixer_CountAllFreeChannels() |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8803 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8804 ALuint retval; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8805 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8806 SDL_LockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8807 #endif |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8808 retval = Internal_CountAllFreeChannels(); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8809 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8810 SDL_UnlockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8811 #endif |
0 | 8812 return retval; |
8813 } | |
8814 | |
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8815 ALuint ALmixer_CountUnreservedFreeChannels() |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8816 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8817 ALuint retval; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8818 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8819 SDL_LockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8820 #endif |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8821 retval = Internal_CountUnreservedFreeChannels(); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8822 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8823 SDL_UnlockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8824 #endif |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8825 return retval; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8826 } |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8827 |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8828 ALuint ALmixer_CountAllUsedChannels() |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8829 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8830 ALuint retval; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8831 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8832 SDL_LockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8833 #endif |
0 | 8834 retval = Internal_CountAllUsedChannels(); |
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8835 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8836 SDL_UnlockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8837 #endif |
0 | 8838 return retval; |
8839 } | |
8840 | |
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8841 ALuint ALmixer_CountUnreservedUsedChannels() |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8842 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8843 ALuint retval; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8844 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8845 SDL_LockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8846 #endif |
0 | 8847 retval = Internal_CountUnreservedUsedChannels(); |
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8848 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8849 SDL_UnlockMutex(s_simpleLock); |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8850 #endif |
0 | 8851 return retval; |
8852 } | |
8853 | |
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8854 ALboolean ALmixer_IsPredecoded(ALmixer_Data* data) |
1 | 8855 { |
8856 if(NULL == data) | |
8857 { | |
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8858 return AL_FALSE; |
1 | 8859 } |
8860 return data->decoded_all; | |
8861 } | |
8862 | |
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8863 ALboolean ALmixer_CompiledWithThreadBackend() |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8864 { |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8865 #ifdef ENABLE_ALMIXER_THREADS |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8866 return AL_TRUE; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8867 #else |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8868 return AL_FALSE; |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8869 #endif |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8870 } |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8871 |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8872 |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8873 |
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8874 |