Mercurial > mplayer.hg
annotate libao2/ao_sdl.c @ 36802:621b61818a0f
configure: add -Zhigh-mem linker flag for OS/2
x264 encoder crashes due to memory allocation failure. This fixes the problem.
author | komh |
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date | Thu, 20 Feb 2014 13:59:34 +0000 |
parents | 0bce77ccae9a |
children |
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 * |
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6 * This file is part of MPlayer. |
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7 * |
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8 * MPlayer is free software; you can redistribute it and/or modify |
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9 * it under the terms of the GNU General Public License as published by |
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10 * the Free Software Foundation; either version 2 of the License, or |
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11 * (at your option) any later version. |
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12 * |
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13 * MPlayer is distributed in the hope that it will be useful, |
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14 * but WITHOUT ANY WARRANTY; without even the implied warranty of |
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15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the |
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16 * GNU General Public License for more details. |
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17 * |
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18 * You should have received a copy of the GNU General Public License along |
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19 * along with MPlayer; if not, write to the Free Software |
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20 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
972 | 21 */ |
22 | |
966 | 23 #include <stdio.h> |
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24 #include <stdlib.h> |
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25 #include <string.h> |
966 | 26 |
14123 | 27 #include "config.h" |
28 #include "mp_msg.h" | |
29 #include "help_mp.h" | |
8027 | 30 |
966 | 31 #include "audio_out.h" |
32 #include "audio_out_internal.h" | |
14245 | 33 #include "libaf/af_format.h" |
30136 | 34 #ifdef CONFIG_SDL_SDL_H |
35 #include <SDL/SDL.h> | |
36 #else | |
6184 | 37 #include <SDL.h> |
30136 | 38 #endif |
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39 #include "osdep/timer.h" |
34645 | 40 #include "osdep/setenv.h" |
966 | 41 |
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42 #include "libavutil/fifo.h" |
972 | 43 |
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44 static const ao_info_t info = |
966 | 45 { |
46 "SDLlib audio output", | |
47 "sdl", | |
48 "Felix Buenemann <atmosfear@users.sourceforge.net>", | |
49 "" | |
50 }; | |
51 | |
52 LIBAO_EXTERN(sdl) | |
53 | |
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54 // turn this on if you want to use the slower SDL_MixAudio |
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55 #undef USE_SDL_INTERNAL_MIXER |
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56 |
972 | 57 // Samplesize used by the SDLlib AudioSpec struct |
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58 #if defined(__MINGW32__) || defined(__CYGWIN__) || defined(__AMIGAOS4__) |
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59 #define SAMPLESIZE 2048 |
12113 | 60 #else |
61 #define SAMPLESIZE 1024 | |
62 #endif | |
972 | 63 |
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64 #define CHUNK_SIZE 4096 |
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65 #define NUM_CHUNKS 8 |
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66 #define BUFFSIZE (NUM_CHUNKS * CHUNK_SIZE) |
972 | 67 |
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68 static AVFifoBuffer *buffer; |
972 | 69 |
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70 #ifdef USE_SDL_INTERNAL_MIXER |
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71 static unsigned char volume=SDL_MIX_MAXVOLUME; |
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72 #endif |
972 | 73 |
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74 static int write_buffer(unsigned char* data,int len){ |
29126 | 75 int free = av_fifo_space(buffer); |
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76 if (len > free) len = free; |
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77 return av_fifo_generic_write(buffer, data, len, NULL); |
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78 } |
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79 |
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80 #ifdef USE_SDL_INTERNAL_MIXER |
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81 static void mix_audio(void *dst, void *src, int len) { |
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82 SDL_MixAudio(dst, src, len, volume); |
972 | 83 } |
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84 #endif |
972 | 85 |
86 static int read_buffer(unsigned char* data,int len){ | |
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87 int buffered = av_fifo_size(buffer); |
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88 if (len > buffered) len = buffered; |
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89 #ifdef USE_SDL_INTERNAL_MIXER |
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90 av_fifo_generic_read(buffer, data, len, mix_audio); |
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91 #else |
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92 av_fifo_generic_read(buffer, data, len, NULL); |
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93 #endif |
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94 return len; |
972 | 95 } |
96 | |
97 // end ring buffer stuff | |
966 | 98 |
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99 |
966 | 100 // to set/get/query special features/parameters |
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101 static int control(int cmd,void *arg){ |
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102 #ifdef USE_SDL_INTERNAL_MIXER |
6184 | 103 switch (cmd) { |
104 case AOCONTROL_GET_VOLUME: | |
105 { | |
106 ao_control_vol_t* vol = (ao_control_vol_t*)arg; | |
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107 vol->left = vol->right = volume * 100 / SDL_MIX_MAXVOLUME; |
6184 | 108 return CONTROL_OK; |
109 } | |
110 case AOCONTROL_SET_VOLUME: | |
111 { | |
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112 int diff; |
6184 | 113 ao_control_vol_t* vol = (ao_control_vol_t*)arg; |
114 diff = (vol->left+vol->right) / 2; | |
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115 volume = diff * SDL_MIX_MAXVOLUME / 100; |
6184 | 116 return CONTROL_OK; |
117 } | |
118 } | |
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119 #endif |
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120 return CONTROL_UNKNOWN; |
966 | 121 } |
122 | |
972 | 123 // SDL Callback function |
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124 static void outputaudio(void *unused, Uint8 *stream, int len) |
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125 { |
972 | 126 //SDL_MixAudio(stream, read_buffer(buffers, len), len, SDL_MIX_MAXVOLUME); |
1066 | 127 //if(!full_buffers) printf("SDL: Buffer underrun!\n"); |
128 | |
972 | 129 read_buffer(stream, len); |
1066 | 130 //printf("SDL: Full Buffers: %i\n", full_buffers); |
966 | 131 } |
132 | |
133 // open & setup audio device | |
134 // return: 1=success 0=fail | |
135 static int init(int rate,int channels,int format,int flags){ | |
136 | |
972 | 137 /* SDL Audio Specifications */ |
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138 SDL_AudioSpec aspec, obtained; |
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139 |
972 | 140 /* Allocate ring-buffer memory */ |
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141 buffer = av_fifo_alloc(BUFFSIZE); |
972 | 142 |
14264 | 143 mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_SDL_INFO, rate, (channels > 1) ? "Stereo" : "Mono", af_fmt2str_short(format)); |
983 | 144 |
1189 | 145 if(ao_subdevice) { |
146 setenv("SDL_AUDIODRIVER", ao_subdevice, 1); | |
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147 mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_SDL_DriverInfo, ao_subdevice); |
1189 | 148 } |
3095 | 149 |
7660 | 150 ao_data.channels=channels; |
151 ao_data.samplerate=rate; | |
152 ao_data.format=format; | |
153 | |
3137 | 154 ao_data.bps=channels*rate; |
14245 | 155 if(format != AF_FORMAT_U8 && format != AF_FORMAT_S8) |
3095 | 156 ao_data.bps*=2; |
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157 |
1066 | 158 /* The desired audio format (see SDL_AudioSpec) */ |
159 switch(format) { | |
14245 | 160 case AF_FORMAT_U8: |
1066 | 161 aspec.format = AUDIO_U8; |
162 break; | |
14245 | 163 case AF_FORMAT_S16_LE: |
1066 | 164 aspec.format = AUDIO_S16LSB; |
165 break; | |
14245 | 166 case AF_FORMAT_S16_BE: |
1066 | 167 aspec.format = AUDIO_S16MSB; |
168 break; | |
14245 | 169 case AF_FORMAT_S8: |
1066 | 170 aspec.format = AUDIO_S8; |
171 break; | |
14245 | 172 case AF_FORMAT_U16_LE: |
1066 | 173 aspec.format = AUDIO_U16LSB; |
174 break; | |
14245 | 175 case AF_FORMAT_U16_BE: |
1066 | 176 aspec.format = AUDIO_U16MSB; |
177 break; | |
178 default: | |
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179 aspec.format = AUDIO_S16LSB; |
14245 | 180 ao_data.format = AF_FORMAT_S16_LE; |
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181 mp_msg(MSGT_AO,MSGL_WARN,MSGTR_AO_SDL_UnsupportedAudioFmt, format); |
1066 | 182 } |
183 | |
966 | 184 /* The desired audio frequency in samples-per-second. */ |
185 aspec.freq = rate; | |
186 | |
187 /* Number of channels (mono/stereo) */ | |
188 aspec.channels = channels; | |
189 | |
190 /* 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 | 191 aspec.samples = SAMPLESIZE; |
966 | 192 |
193 /* 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: | |
194 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 | 195 aspec.callback = outputaudio; |
966 | 196 |
197 /* This pointer is passed as the first parameter to the callback function. */ | |
198 aspec.userdata = NULL; | |
199 | |
972 | 200 /* initialize the SDL Audio system */ |
201 if (SDL_Init (SDL_INIT_AUDIO/*|SDL_INIT_NOPARACHUTE*/)) { | |
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202 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_SDL_CantInit, SDL_GetError()); |
972 | 203 return 0; |
204 } | |
205 | |
966 | 206 /* Open the audio device and start playing sound! */ |
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207 if(SDL_OpenAudio(&aspec, &obtained) < 0) { |
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208 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_SDL_CantOpenAudio, SDL_GetError()); |
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209 return 0; |
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210 } |
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211 |
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212 /* did we got what we wanted ? */ |
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213 ao_data.channels=obtained.channels; |
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214 ao_data.samplerate=obtained.freq; |
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215 |
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216 switch(obtained.format) { |
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217 case AUDIO_U8 : |
14245 | 218 ao_data.format = AF_FORMAT_U8; |
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219 break; |
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220 case AUDIO_S16LSB : |
14245 | 221 ao_data.format = AF_FORMAT_S16_LE; |
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222 break; |
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223 case AUDIO_S16MSB : |
14245 | 224 ao_data.format = AF_FORMAT_S16_BE; |
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225 break; |
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226 case AUDIO_S8 : |
14245 | 227 ao_data.format = AF_FORMAT_S8; |
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228 break; |
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229 case AUDIO_U16LSB : |
14245 | 230 ao_data.format = AF_FORMAT_U16_LE; |
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231 break; |
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232 case AUDIO_U16MSB : |
14245 | 233 ao_data.format = AF_FORMAT_U16_BE; |
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234 break; |
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235 default: |
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236 mp_msg(MSGT_AO,MSGL_WARN,MSGTR_AO_SDL_UnsupportedAudioFmt, obtained.format); |
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237 return 0; |
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238 } |
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239 |
8027 | 240 mp_msg(MSGT_AO,MSGL_V,"SDL: buf size = %d\n",obtained.size); |
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241 ao_data.buffersize=obtained.size; |
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242 ao_data.outburst = CHUNK_SIZE; |
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243 |
966 | 244 /* unsilence audio, if callback is ready */ |
245 SDL_PauseAudio(0); | |
246 | |
247 return 1; | |
248 } | |
249 | |
250 // close audio device | |
12145 | 251 static void uninit(int immed){ |
8027 | 252 mp_msg(MSGT_AO,MSGL_V,"SDL: Audio Subsystem shutting down!\n"); |
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253 if (!immed) |
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254 usec_sleep(get_delay() * 1000 * 1000); |
966 | 255 SDL_CloseAudio(); |
972 | 256 SDL_QuitSubSystem(SDL_INIT_AUDIO); |
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257 av_fifo_free(buffer); |
966 | 258 } |
259 | |
260 // stop playing and empty buffers (for seeking/pause) | |
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261 static void reset(void){ |
1066 | 262 |
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263 //printf("SDL: reset called!\n"); |
1066 | 264 |
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265 SDL_PauseAudio(1); |
972 | 266 /* Reset ring-buffer state */ |
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267 av_fifo_reset(buffer); |
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268 SDL_PauseAudio(0); |
966 | 269 } |
270 | |
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271 // stop playing, keep buffers (for pause) |
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272 static void audio_pause(void) |
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273 { |
1070 | 274 |
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275 //printf("SDL: audio_pause called!\n"); |
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276 SDL_PauseAudio(1); |
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277 |
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278 } |
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279 |
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280 // resume playing, after audio_pause() |
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281 static void audio_resume(void) |
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282 { |
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283 //printf("SDL: audio_resume called!\n"); |
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284 SDL_PauseAudio(0); |
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285 } |
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286 |
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287 |
966 | 288 // return: how many bytes can be played without blocking |
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289 static int get_space(void){ |
29126 | 290 return av_fifo_space(buffer); |
966 | 291 } |
292 | |
293 // plays 'len' bytes of 'data' | |
294 // it should round it down to outburst*n | |
295 // return: number of bytes played | |
296 static int play(void* data,int len,int flags){ | |
972 | 297 |
22104 | 298 if (!(flags & AOPLAY_FINAL_CHUNK)) |
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299 len = (len/ao_data.outburst)*ao_data.outburst; |
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300 #if 0 |
972 | 301 int ret; |
966 | 302 |
972 | 303 /* Audio locking prohibits call of outputaudio */ |
966 | 304 SDL_LockAudio(); |
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305 // copy audio stream into ring-buffer |
972 | 306 ret = write_buffer(data, len); |
966 | 307 SDL_UnlockAudio(); |
308 | |
972 | 309 return ret; |
310 #else | |
311 return write_buffer(data, len); | |
312 #endif | |
966 | 313 } |
314 | |
3095 | 315 // return: delay in seconds between first and last sample in buffer |
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316 static float get_delay(void){ |
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317 int buffered = av_fifo_size(buffer); // could be less |
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318 return (float)(buffered + ao_data.buffersize)/(float)ao_data.bps; |
966 | 319 } |