Mercurial > mplayer.hg
annotate libao2/ao_sdl.c @ 27594:2a014f30d031
Update the copyright statement.
author | syrjala |
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date | Mon, 15 Sep 2008 20:46:43 +0000 |
parents | d97a607821f1 |
children | 48c1ae64255b |
rev | line source |
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1 /* |
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2 * SDLlib audio output driver for MPlayer |
972 | 3 * |
23734 | 4 * Copyleft 2001 by Felix BĆ¼nemann (atmosfear@users.sf.net) |
972 | 5 * |
6 * Thanks to Arpi for nice ringbuffer-code! | |
7 * | |
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8 * This file is part of MPlayer. |
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9 * |
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10 * MPlayer is free software; you can redistribute it and/or modify |
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11 * it under the terms of the GNU General Public License as published by |
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12 * the Free Software Foundation; either version 2 of the License, or |
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13 * (at your option) any later version. |
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14 * |
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15 * MPlayer is distributed in the hope that it will be useful, |
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16 * but WITHOUT ANY WARRANTY; without even the implied warranty of |
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17 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the |
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18 * GNU General Public License for more details. |
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19 * |
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20 * You should have received a copy of the GNU General Public License along |
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21 * along with MPlayer; if not, write to the Free Software |
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22 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
972 | 23 */ |
24 | |
966 | 25 #include <stdio.h> |
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26 #include <stdlib.h> |
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27 #include <string.h> |
966 | 28 |
14123 | 29 #include "config.h" |
30 #include "mp_msg.h" | |
31 #include "help_mp.h" | |
8027 | 32 |
966 | 33 #include "audio_out.h" |
34 #include "audio_out_internal.h" | |
14245 | 35 #include "libaf/af_format.h" |
6184 | 36 #include <SDL.h> |
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37 #include "osdep/timer.h" |
966 | 38 |
14123 | 39 #include "libvo/fastmemcpy.h" |
972 | 40 |
966 | 41 static ao_info_t info = |
42 { | |
43 "SDLlib audio output", | |
44 "sdl", | |
45 "Felix Buenemann <atmosfear@users.sourceforge.net>", | |
46 "" | |
47 }; | |
48 | |
49 LIBAO_EXTERN(sdl) | |
50 | |
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51 // turn this on if you want to use the slower SDL_MixAudio |
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52 #undef USE_SDL_INTERNAL_MIXER |
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53 |
972 | 54 // Samplesize used by the SDLlib AudioSpec struct |
25335 | 55 #if defined(WIN32) || defined(__AMIGAOS4__) |
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56 #define SAMPLESIZE 2048 |
12113 | 57 #else |
58 #define SAMPLESIZE 1024 | |
59 #endif | |
972 | 60 |
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61 #define CHUNK_SIZE 4096 |
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62 #define NUM_CHUNKS 8 |
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63 // This type of ring buffer may never fill up completely, at least |
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64 // one byte must always be unused. |
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65 // For performance reasons (alignment etc.) one whole chunk always stays |
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66 // empty, not only one byte. |
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67 #define BUFFSIZE ((NUM_CHUNKS + 1) * CHUNK_SIZE) |
972 | 68 |
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69 static unsigned char *buffer; |
972 | 70 |
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71 // may only be modified by SDL's playback thread or while it is stopped |
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72 static volatile int read_pos; |
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73 // may only be modified by mplayer's thread |
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74 static volatile int write_pos; |
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75 #ifdef USE_SDL_INTERNAL_MIXER |
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76 static unsigned char volume=SDL_MIX_MAXVOLUME; |
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77 #endif |
972 | 78 |
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79 // may only be called by mplayer's thread |
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80 // return value may change between immediately following two calls, |
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81 // and the real number of free bytes might be larger! |
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82 static int buf_free(void) { |
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83 int free = read_pos - write_pos - CHUNK_SIZE; |
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84 if (free < 0) free += BUFFSIZE; |
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85 return free; |
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86 } |
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87 |
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88 // may only be called by SDL's playback thread |
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89 // return value may change between immediately following two calls, |
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90 // and the real number of buffered bytes might be larger! |
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91 static int buf_used(void) { |
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92 int used = write_pos - read_pos; |
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93 if (used < 0) used += BUFFSIZE; |
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94 return used; |
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95 } |
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96 |
972 | 97 static int write_buffer(unsigned char* data,int len){ |
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98 int first_len = BUFFSIZE - write_pos; |
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99 int free = buf_free(); |
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100 if (len > free) len = free; |
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101 if (first_len > len) first_len = len; |
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102 // till end of buffer |
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103 fast_memcpy (&buffer[write_pos], data, first_len); |
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104 if (len > first_len) { // we have to wrap around |
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105 // remaining part from beginning of buffer |
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106 fast_memcpy (buffer, &data[first_len], len - first_len); |
972 | 107 } |
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108 write_pos = (write_pos + len) % BUFFSIZE; |
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109 return len; |
972 | 110 } |
111 | |
112 static int read_buffer(unsigned char* data,int len){ | |
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113 int first_len = BUFFSIZE - read_pos; |
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114 int buffered = buf_used(); |
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115 if (len > buffered) len = buffered; |
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116 if (first_len > len) first_len = len; |
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117 // till end of buffer |
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118 #ifdef USE_SDL_INTERNAL_MIXER |
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119 SDL_MixAudio (data, &buffer[read_pos], first_len, volume); |
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120 #else |
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121 fast_memcpy (data, &buffer[read_pos], first_len); |
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122 #endif |
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123 if (len > first_len) { // we have to wrap around |
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124 // remaining part from beginning of buffer |
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125 #ifdef USE_SDL_INTERNAL_MIXER |
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126 SDL_MixAudio (&data[first_len], buffer, len - first_len, volume); |
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127 #else |
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128 fast_memcpy (&data[first_len], buffer, len - first_len); |
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129 #endif |
972 | 130 } |
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131 read_pos = (read_pos + len) % BUFFSIZE; |
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132 return len; |
972 | 133 } |
134 | |
135 // end ring buffer stuff | |
966 | 136 |
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137 |
966 | 138 // to set/get/query special features/parameters |
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139 static int control(int cmd,void *arg){ |
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140 #ifdef USE_SDL_INTERNAL_MIXER |
6184 | 141 switch (cmd) { |
142 case AOCONTROL_GET_VOLUME: | |
143 { | |
144 ao_control_vol_t* vol = (ao_control_vol_t*)arg; | |
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145 vol->left = vol->right = volume * 100 / SDL_MIX_MAXVOLUME; |
6184 | 146 return CONTROL_OK; |
147 } | |
148 case AOCONTROL_SET_VOLUME: | |
149 { | |
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150 int diff; |
6184 | 151 ao_control_vol_t* vol = (ao_control_vol_t*)arg; |
152 diff = (vol->left+vol->right) / 2; | |
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153 volume = diff * SDL_MIX_MAXVOLUME / 100; |
6184 | 154 return CONTROL_OK; |
155 } | |
156 } | |
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157 #endif |
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158 return CONTROL_UNKNOWN; |
966 | 159 } |
160 | |
972 | 161 // SDL Callback function |
162 void outputaudio(void *unused, Uint8 *stream, int len) { | |
163 //SDL_MixAudio(stream, read_buffer(buffers, len), len, SDL_MIX_MAXVOLUME); | |
1066 | 164 //if(!full_buffers) printf("SDL: Buffer underrun!\n"); |
165 | |
972 | 166 read_buffer(stream, len); |
1066 | 167 //printf("SDL: Full Buffers: %i\n", full_buffers); |
966 | 168 } |
169 | |
170 // open & setup audio device | |
171 // return: 1=success 0=fail | |
172 static int init(int rate,int channels,int format,int flags){ | |
173 | |
972 | 174 /* SDL Audio Specifications */ |
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175 SDL_AudioSpec aspec, obtained; |
972 | 176 |
177 /* Allocate ring-buffer memory */ | |
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178 buffer = (unsigned char *) malloc(BUFFSIZE); |
972 | 179 |
14264 | 180 mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_SDL_INFO, rate, (channels > 1) ? "Stereo" : "Mono", af_fmt2str_short(format)); |
983 | 181 |
1189 | 182 if(ao_subdevice) { |
183 setenv("SDL_AUDIODRIVER", ao_subdevice, 1); | |
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184 mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_SDL_DriverInfo, ao_subdevice); |
1189 | 185 } |
3095 | 186 |
7660 | 187 ao_data.channels=channels; |
188 ao_data.samplerate=rate; | |
189 ao_data.format=format; | |
190 | |
3137 | 191 ao_data.bps=channels*rate; |
14245 | 192 if(format != AF_FORMAT_U8 && format != AF_FORMAT_S8) |
3095 | 193 ao_data.bps*=2; |
966 | 194 |
1066 | 195 /* The desired audio format (see SDL_AudioSpec) */ |
196 switch(format) { | |
14245 | 197 case AF_FORMAT_U8: |
1066 | 198 aspec.format = AUDIO_U8; |
199 break; | |
14245 | 200 case AF_FORMAT_S16_LE: |
1066 | 201 aspec.format = AUDIO_S16LSB; |
202 break; | |
14245 | 203 case AF_FORMAT_S16_BE: |
1066 | 204 aspec.format = AUDIO_S16MSB; |
205 break; | |
14245 | 206 case AF_FORMAT_S8: |
1066 | 207 aspec.format = AUDIO_S8; |
208 break; | |
14245 | 209 case AF_FORMAT_U16_LE: |
1066 | 210 aspec.format = AUDIO_U16LSB; |
211 break; | |
14245 | 212 case AF_FORMAT_U16_BE: |
1066 | 213 aspec.format = AUDIO_U16MSB; |
214 break; | |
215 default: | |
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216 aspec.format = AUDIO_S16LSB; |
14245 | 217 ao_data.format = AF_FORMAT_S16_LE; |
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218 mp_msg(MSGT_AO,MSGL_WARN,MSGTR_AO_SDL_UnsupportedAudioFmt, format); |
1066 | 219 } |
220 | |
966 | 221 /* The desired audio frequency in samples-per-second. */ |
222 aspec.freq = rate; | |
223 | |
224 /* Number of channels (mono/stereo) */ | |
225 aspec.channels = channels; | |
226 | |
227 /* 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 | 228 aspec.samples = SAMPLESIZE; |
966 | 229 |
230 /* 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: | |
231 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 | 232 aspec.callback = outputaudio; |
966 | 233 |
234 /* This pointer is passed as the first parameter to the callback function. */ | |
235 aspec.userdata = NULL; | |
236 | |
972 | 237 /* initialize the SDL Audio system */ |
238 if (SDL_Init (SDL_INIT_AUDIO/*|SDL_INIT_NOPARACHUTE*/)) { | |
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239 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_SDL_CantInit, SDL_GetError()); |
972 | 240 return 0; |
241 } | |
242 | |
966 | 243 /* Open the audio device and start playing sound! */ |
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244 if(SDL_OpenAudio(&aspec, &obtained) < 0) { |
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245 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_SDL_CantOpenAudio, SDL_GetError()); |
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246 return 0; |
966 | 247 } |
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248 |
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249 /* did we got what we wanted ? */ |
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250 ao_data.channels=obtained.channels; |
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251 ao_data.samplerate=obtained.freq; |
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252 |
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253 switch(obtained.format) { |
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254 case AUDIO_U8 : |
14245 | 255 ao_data.format = AF_FORMAT_U8; |
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256 break; |
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257 case AUDIO_S16LSB : |
14245 | 258 ao_data.format = AF_FORMAT_S16_LE; |
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259 break; |
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260 case AUDIO_S16MSB : |
14245 | 261 ao_data.format = AF_FORMAT_S16_BE; |
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262 break; |
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263 case AUDIO_S8 : |
14245 | 264 ao_data.format = AF_FORMAT_S8; |
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265 break; |
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266 case AUDIO_U16LSB : |
14245 | 267 ao_data.format = AF_FORMAT_U16_LE; |
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268 break; |
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269 case AUDIO_U16MSB : |
14245 | 270 ao_data.format = AF_FORMAT_U16_BE; |
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271 break; |
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272 default: |
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273 mp_msg(MSGT_AO,MSGL_WARN,MSGTR_AO_SDL_UnsupportedAudioFmt, obtained.format); |
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274 return 0; |
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275 } |
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276 |
8027 | 277 mp_msg(MSGT_AO,MSGL_V,"SDL: buf size = %d\n",obtained.size); |
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278 ao_data.buffersize=obtained.size; |
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279 ao_data.outburst = CHUNK_SIZE; |
974 | 280 |
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281 reset(); |
966 | 282 /* unsilence audio, if callback is ready */ |
283 SDL_PauseAudio(0); | |
284 | |
285 return 1; | |
286 } | |
287 | |
288 // close audio device | |
12145 | 289 static void uninit(int immed){ |
8027 | 290 mp_msg(MSGT_AO,MSGL_V,"SDL: Audio Subsystem shutting down!\n"); |
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291 if (!immed) |
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292 usec_sleep(get_delay() * 1000 * 1000); |
966 | 293 SDL_CloseAudio(); |
972 | 294 SDL_QuitSubSystem(SDL_INIT_AUDIO); |
966 | 295 } |
296 | |
297 // stop playing and empty buffers (for seeking/pause) | |
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298 static void reset(void){ |
1066 | 299 |
1091 | 300 //printf("SDL: reset called!\n"); |
1066 | 301 |
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302 SDL_PauseAudio(1); |
972 | 303 /* Reset ring-buffer state */ |
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304 read_pos = 0; |
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305 write_pos = 0; |
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306 SDL_PauseAudio(0); |
966 | 307 } |
308 | |
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309 // stop playing, keep buffers (for pause) |
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310 static void audio_pause(void) |
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311 { |
1070 | 312 |
1091 | 313 //printf("SDL: audio_pause called!\n"); |
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314 SDL_PauseAudio(1); |
1070 | 315 |
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316 } |
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317 |
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318 // resume playing, after audio_pause() |
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319 static void audio_resume(void) |
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320 { |
1091 | 321 //printf("SDL: audio_resume called!\n"); |
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322 SDL_PauseAudio(0); |
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323 } |
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324 |
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325 |
966 | 326 // return: how many bytes can be played without blocking |
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327 static int get_space(void){ |
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328 return buf_free(); |
966 | 329 } |
330 | |
331 // plays 'len' bytes of 'data' | |
332 // it should round it down to outburst*n | |
333 // return: number of bytes played | |
334 static int play(void* data,int len,int flags){ | |
972 | 335 |
22104 | 336 if (!(flags & AOPLAY_FINAL_CHUNK)) |
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337 len = (len/ao_data.outburst)*ao_data.outburst; |
985 | 338 #if 0 |
972 | 339 int ret; |
966 | 340 |
972 | 341 /* Audio locking prohibits call of outputaudio */ |
966 | 342 SDL_LockAudio(); |
972 | 343 // copy audio stream into ring-buffer |
344 ret = write_buffer(data, len); | |
966 | 345 SDL_UnlockAudio(); |
346 | |
972 | 347 return ret; |
348 #else | |
349 return write_buffer(data, len); | |
350 #endif | |
966 | 351 } |
352 | |
3095 | 353 // return: delay in seconds between first and last sample in buffer |
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354 static float get_delay(void){ |
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355 int buffered = BUFFSIZE - CHUNK_SIZE - buf_free(); // could be less |
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356 return (float)(buffered + ao_data.buffersize)/(float)ao_data.bps; |
966 | 357 } |
358 | |
359 | |
360 | |
361 | |
362 | |
363 |