Mercurial > mplayer.hg
annotate libao2/ao_sdl.c @ 20962:a95ed9a69caf
Put files fully owned by my under GPL v2 "or later"
author | reimar |
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date | Fri, 17 Nov 2006 10:03:33 +0000 |
parents | f580a7755ac5 |
children | cc6dc96035b8 |
rev | line source |
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972 | 1 /* |
2 * ao_sdl.c - libao2 SDLlib Audio Output Driver for MPlayer | |
3 * | |
4 * This driver is under the same license as MPlayer. | |
11750 | 5 * (http://www.mplayerhq.hu) |
972 | 6 * |
7 * Copyleft 2001 by Felix Bünemann (atmosfear@users.sf.net) | |
8 * | |
9 * Thanks to Arpi for nice ringbuffer-code! | |
10 * | |
11 */ | |
12 | |
966 | 13 #include <stdio.h> |
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14 #include <stdlib.h> |
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15 #include <string.h> |
966 | 16 |
14123 | 17 #include "config.h" |
18 #include "mp_msg.h" | |
19 #include "help_mp.h" | |
8027 | 20 |
966 | 21 #include "audio_out.h" |
22 #include "audio_out_internal.h" | |
14245 | 23 #include "libaf/af_format.h" |
6184 | 24 #include <SDL.h> |
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25 #include "osdep/timer.h" |
966 | 26 |
14123 | 27 #include "libvo/fastmemcpy.h" |
972 | 28 |
966 | 29 static ao_info_t info = |
30 { | |
31 "SDLlib audio output", | |
32 "sdl", | |
33 "Felix Buenemann <atmosfear@users.sourceforge.net>", | |
34 "" | |
35 }; | |
36 | |
37 LIBAO_EXTERN(sdl) | |
38 | |
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39 // turn this on if you want to use the slower SDL_MixAudio |
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40 #undef USE_SDL_INTERNAL_MIXER |
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41 |
972 | 42 // Samplesize used by the SDLlib AudioSpec struct |
12113 | 43 #ifdef WIN32 |
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44 #define SAMPLESIZE 2048 |
12113 | 45 #else |
46 #define SAMPLESIZE 1024 | |
47 #endif | |
972 | 48 |
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49 #define CHUNK_SIZE 4096 |
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50 #define NUM_CHUNKS 8 |
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51 // This type of ring buffer may never fill up completely, at least |
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52 // one byte must always be unused. |
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53 // For performance reasons (alignment etc.) one whole chunk always stays |
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54 // empty, not only one byte. |
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55 #define BUFFSIZE ((NUM_CHUNKS + 1) * CHUNK_SIZE) |
972 | 56 |
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57 static unsigned char *buffer; |
972 | 58 |
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59 // may only be modified by SDL's playback thread or while it is stopped |
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60 static volatile int read_pos; |
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61 // may only be modified by mplayer's thread |
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62 static volatile int write_pos; |
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63 #ifdef USE_SDL_INTERNAL_MIXER |
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64 static unsigned char volume=SDL_MIX_MAXVOLUME; |
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65 #endif |
972 | 66 |
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67 // may only be called by mplayer's thread |
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68 // return value may change between immediately following two calls, |
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69 // and the real number of free bytes might be larger! |
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70 static int buf_free(void) { |
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71 int free = read_pos - write_pos - CHUNK_SIZE; |
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72 if (free < 0) free += BUFFSIZE; |
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73 return free; |
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74 } |
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75 |
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76 // may only be called by SDL's playback thread |
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77 // return value may change between immediately following two calls, |
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78 // and the real number of buffered bytes might be larger! |
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79 static int buf_used(void) { |
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80 int used = write_pos - read_pos; |
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81 if (used < 0) used += BUFFSIZE; |
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82 return used; |
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83 } |
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84 |
972 | 85 static int write_buffer(unsigned char* data,int len){ |
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86 int first_len = BUFFSIZE - write_pos; |
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87 int free = buf_free(); |
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88 if (len > free) len = free; |
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89 if (first_len > len) first_len = len; |
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90 // till end of buffer |
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91 memcpy (&buffer[write_pos], data, first_len); |
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92 if (len > first_len) { // we have to wrap around |
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93 // remaining part from beginning of buffer |
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94 memcpy (buffer, &data[first_len], len - first_len); |
972 | 95 } |
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96 write_pos = (write_pos + len) % BUFFSIZE; |
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97 return len; |
972 | 98 } |
99 | |
100 static int read_buffer(unsigned char* data,int len){ | |
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101 int first_len = BUFFSIZE - read_pos; |
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102 int buffered = buf_used(); |
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103 if (len > buffered) len = buffered; |
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104 if (first_len > len) first_len = len; |
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105 // till end of buffer |
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106 #ifdef USE_SDL_INTERNAL_MIXER |
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107 SDL_MixAudio (data, &buffer[read_pos], first_len, volume); |
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108 #else |
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109 memcpy (data, &buffer[read_pos], first_len); |
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110 #endif |
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111 if (len > first_len) { // we have to wrap around |
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112 // remaining part from beginning of buffer |
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113 #ifdef USE_SDL_INTERNAL_MIXER |
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114 SDL_MixAudio (&data[first_len], buffer, len - first_len, volume); |
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115 #else |
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116 memcpy (&data[first_len], buffer, len - first_len); |
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117 #endif |
972 | 118 } |
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119 read_pos = (read_pos + len) % BUFFSIZE; |
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120 return len; |
972 | 121 } |
122 | |
123 // end ring buffer stuff | |
966 | 124 |
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125 |
966 | 126 // to set/get/query special features/parameters |
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127 static int control(int cmd,void *arg){ |
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128 #ifdef USE_SDL_INTERNAL_MIXER |
6184 | 129 switch (cmd) { |
130 case AOCONTROL_GET_VOLUME: | |
131 { | |
132 ao_control_vol_t* vol = (ao_control_vol_t*)arg; | |
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133 vol->left = vol->right = volume * 100 / SDL_MIX_MAXVOLUME; |
6184 | 134 return CONTROL_OK; |
135 } | |
136 case AOCONTROL_SET_VOLUME: | |
137 { | |
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138 int diff; |
6184 | 139 ao_control_vol_t* vol = (ao_control_vol_t*)arg; |
140 diff = (vol->left+vol->right) / 2; | |
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141 volume = diff * SDL_MIX_MAXVOLUME / 100; |
6184 | 142 return CONTROL_OK; |
143 } | |
144 } | |
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145 #endif |
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146 return CONTROL_UNKNOWN; |
966 | 147 } |
148 | |
972 | 149 // SDL Callback function |
150 void outputaudio(void *unused, Uint8 *stream, int len) { | |
151 //SDL_MixAudio(stream, read_buffer(buffers, len), len, SDL_MIX_MAXVOLUME); | |
1066 | 152 //if(!full_buffers) printf("SDL: Buffer underrun!\n"); |
153 | |
972 | 154 read_buffer(stream, len); |
1066 | 155 //printf("SDL: Full Buffers: %i\n", full_buffers); |
966 | 156 } |
157 | |
158 // open & setup audio device | |
159 // return: 1=success 0=fail | |
160 static int init(int rate,int channels,int format,int flags){ | |
161 | |
972 | 162 /* SDL Audio Specifications */ |
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163 SDL_AudioSpec aspec, obtained; |
972 | 164 |
165 /* Allocate ring-buffer memory */ | |
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166 buffer = (unsigned char *) malloc(BUFFSIZE); |
972 | 167 |
14264 | 168 mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_SDL_INFO, rate, (channels > 1) ? "Stereo" : "Mono", af_fmt2str_short(format)); |
983 | 169 |
1189 | 170 if(ao_subdevice) { |
171 setenv("SDL_AUDIODRIVER", ao_subdevice, 1); | |
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172 mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_SDL_DriverInfo, ao_subdevice); |
1189 | 173 } |
3095 | 174 |
7660 | 175 ao_data.channels=channels; |
176 ao_data.samplerate=rate; | |
177 ao_data.format=format; | |
178 | |
3137 | 179 ao_data.bps=channels*rate; |
14245 | 180 if(format != AF_FORMAT_U8 && format != AF_FORMAT_S8) |
3095 | 181 ao_data.bps*=2; |
966 | 182 |
1066 | 183 /* The desired audio format (see SDL_AudioSpec) */ |
184 switch(format) { | |
14245 | 185 case AF_FORMAT_U8: |
1066 | 186 aspec.format = AUDIO_U8; |
187 break; | |
14245 | 188 case AF_FORMAT_S16_LE: |
1066 | 189 aspec.format = AUDIO_S16LSB; |
190 break; | |
14245 | 191 case AF_FORMAT_S16_BE: |
1066 | 192 aspec.format = AUDIO_S16MSB; |
193 break; | |
14245 | 194 case AF_FORMAT_S8: |
1066 | 195 aspec.format = AUDIO_S8; |
196 break; | |
14245 | 197 case AF_FORMAT_U16_LE: |
1066 | 198 aspec.format = AUDIO_U16LSB; |
199 break; | |
14245 | 200 case AF_FORMAT_U16_BE: |
1066 | 201 aspec.format = AUDIO_U16MSB; |
202 break; | |
203 default: | |
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204 aspec.format = AUDIO_S16LSB; |
14245 | 205 ao_data.format = AF_FORMAT_S16_LE; |
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206 mp_msg(MSGT_AO,MSGL_WARN,MSGTR_AO_SDL_UnsupportedAudioFmt, format); |
1066 | 207 } |
208 | |
966 | 209 /* The desired audio frequency in samples-per-second. */ |
210 aspec.freq = rate; | |
211 | |
212 /* Number of channels (mono/stereo) */ | |
213 aspec.channels = channels; | |
214 | |
215 /* The desired size of the audio buffer in samples. This number should be a power of two, and may be adjusted by the audio driver to a value more suitable for the hardware. Good values seem to range between 512 and 8192 inclusive, depending on the application and CPU speed. Smaller values yield faster response time, but can lead to underflow if the application is doing heavy processing and cannot fill the audio buffer in time. A stereo sample consists of both right and left channels in LR ordering. Note that the number of samples is directly related to time by the following formula: ms = (samples*1000)/freq */ | |
972 | 216 aspec.samples = SAMPLESIZE; |
966 | 217 |
218 /* This should be set to a function that will be called when the audio device is ready for more data. It is passed a pointer to the audio buffer, and the length in bytes of the audio buffer. This function usually runs in a separate thread, and so you should protect data structures that it accesses by calling SDL_LockAudio and SDL_UnlockAudio in your code. The callback prototype is: | |
219 void callback(void *userdata, Uint8 *stream, int len); userdata is the pointer stored in userdata field of the SDL_AudioSpec. stream is a pointer to the audio buffer you want to fill with information and len is the length of the audio buffer in bytes. */ | |
972 | 220 aspec.callback = outputaudio; |
966 | 221 |
222 /* This pointer is passed as the first parameter to the callback function. */ | |
223 aspec.userdata = NULL; | |
224 | |
972 | 225 /* initialize the SDL Audio system */ |
226 if (SDL_Init (SDL_INIT_AUDIO/*|SDL_INIT_NOPARACHUTE*/)) { | |
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227 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_SDL_CantInit, SDL_GetError()); |
972 | 228 return 0; |
229 } | |
230 | |
966 | 231 /* Open the audio device and start playing sound! */ |
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232 if(SDL_OpenAudio(&aspec, &obtained) < 0) { |
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233 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_SDL_CantOpenAudio, SDL_GetError()); |
966 | 234 return(0); |
235 } | |
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236 |
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237 /* did we got what we wanted ? */ |
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238 ao_data.channels=obtained.channels; |
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239 ao_data.samplerate=obtained.freq; |
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240 |
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241 switch(obtained.format) { |
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242 case AUDIO_U8 : |
14245 | 243 ao_data.format = AF_FORMAT_U8; |
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244 break; |
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245 case AUDIO_S16LSB : |
14245 | 246 ao_data.format = AF_FORMAT_S16_LE; |
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247 break; |
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248 case AUDIO_S16MSB : |
14245 | 249 ao_data.format = AF_FORMAT_S16_BE; |
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250 break; |
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251 case AUDIO_S8 : |
14245 | 252 ao_data.format = AF_FORMAT_S8; |
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253 break; |
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254 case AUDIO_U16LSB : |
14245 | 255 ao_data.format = AF_FORMAT_U16_LE; |
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256 break; |
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257 case AUDIO_U16MSB : |
14245 | 258 ao_data.format = AF_FORMAT_U16_BE; |
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259 break; |
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260 default: |
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261 mp_msg(MSGT_AO,MSGL_WARN,MSGTR_AO_SDL_UnsupportedAudioFmt, obtained.format); |
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262 return 0; |
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263 } |
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264 |
8027 | 265 mp_msg(MSGT_AO,MSGL_V,"SDL: buf size = %d\n",obtained.size); |
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266 ao_data.buffersize=obtained.size; |
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267 ao_data.outburst = CHUNK_SIZE; |
974 | 268 |
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269 reset(); |
966 | 270 /* unsilence audio, if callback is ready */ |
271 SDL_PauseAudio(0); | |
272 | |
273 return 1; | |
274 } | |
275 | |
276 // close audio device | |
12145 | 277 static void uninit(int immed){ |
8027 | 278 mp_msg(MSGT_AO,MSGL_V,"SDL: Audio Subsystem shutting down!\n"); |
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279 if (!immed) |
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280 usec_sleep(get_delay() * 1000 * 1000); |
966 | 281 SDL_CloseAudio(); |
972 | 282 SDL_QuitSubSystem(SDL_INIT_AUDIO); |
966 | 283 } |
284 | |
285 // stop playing and empty buffers (for seeking/pause) | |
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286 static void reset(void){ |
1066 | 287 |
1091 | 288 //printf("SDL: reset called!\n"); |
1066 | 289 |
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290 SDL_PauseAudio(1); |
972 | 291 /* Reset ring-buffer state */ |
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292 read_pos = 0; |
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293 write_pos = 0; |
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294 SDL_PauseAudio(0); |
966 | 295 } |
296 | |
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297 // stop playing, keep buffers (for pause) |
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298 static void audio_pause(void) |
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299 { |
1070 | 300 |
1091 | 301 //printf("SDL: audio_pause called!\n"); |
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302 SDL_PauseAudio(1); |
1070 | 303 |
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304 } |
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305 |
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306 // resume playing, after audio_pause() |
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307 static void audio_resume(void) |
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308 { |
1091 | 309 //printf("SDL: audio_resume called!\n"); |
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310 SDL_PauseAudio(0); |
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311 } |
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312 |
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313 |
966 | 314 // return: how many bytes can be played without blocking |
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315 static int get_space(void){ |
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316 return buf_free(); |
966 | 317 } |
318 | |
319 // plays 'len' bytes of 'data' | |
320 // it should round it down to outburst*n | |
321 // return: number of bytes played | |
322 static int play(void* data,int len,int flags){ | |
972 | 323 |
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324 len = (len/ao_data.outburst)*ao_data.outburst; |
985 | 325 #if 0 |
972 | 326 int ret; |
966 | 327 |
972 | 328 /* Audio locking prohibits call of outputaudio */ |
966 | 329 SDL_LockAudio(); |
972 | 330 // copy audio stream into ring-buffer |
331 ret = write_buffer(data, len); | |
966 | 332 SDL_UnlockAudio(); |
333 | |
972 | 334 return ret; |
335 #else | |
336 return write_buffer(data, len); | |
337 #endif | |
966 | 338 } |
339 | |
3095 | 340 // return: delay in seconds between first and last sample in buffer |
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341 static float get_delay(void){ |
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342 int buffered = BUFFSIZE - CHUNK_SIZE - buf_free(); // could be less |
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343 return (float)(buffered + ao_data.buffersize)/(float)ao_data.bps; |
966 | 344 } |
345 | |
346 | |
347 | |
348 | |
349 | |
350 |