Mercurial > mplayer.hg
annotate libao2/ao_win32.c @ 18888:ab29b3c4f94e
parse_cfgfiles as static in mplayer.c/mencoder.c, patch by Stefan Huehner, stefan AT huehner-org
author | reynaldo |
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date | Mon, 03 Jul 2006 23:18:06 +0000 |
parents | cc9c088305d9 |
children | 99e20a22d5d0 |
rev | line source |
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7913 | 1 /****************************************************************************** |
2 * ao_win32.c: Windows waveOut interface for MPlayer | |
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3 * Copyright (c) 2002 - 2004 Sascha Sommer <saschasommer@freenet.de>. |
7913 | 4 * |
5 * This program is free software; you can redistribute it and/or modify | |
6 * it under the terms of the GNU General Public License as published by | |
7 * the Free Software Foundation; either version 2 of the License, or | |
8 * (at your option) any later version. | |
9 * | |
10 * This program is distributed in the hope that it will be useful, | |
11 * but WITHOUT ANY WARRANTY; without even the implied warranty of | |
12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the | |
13 * GNU General Public License for more details. | |
14 * | |
15 * You should have received a copy of the GNU General Public License | |
16 * along with this program; if not, write to the Free Software | |
17 * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111, USA. | |
18 * | |
19 *****************************************************************************/ | |
20 | |
21 #include <stdio.h> | |
22 #include <stdlib.h> | |
23 #include <windows.h> | |
24 #include <mmsystem.h> | |
25 | |
14479 | 26 #include "config.h" |
14245 | 27 #include "libaf/af_format.h" |
7913 | 28 #include "audio_out.h" |
29 #include "audio_out_internal.h" | |
14123 | 30 #include "mp_msg.h" |
31 #include "libvo/fastmemcpy.h" | |
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32 #include "osdep/timer.h" |
7913 | 33 |
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34 #define WAVE_FORMAT_DOLBY_AC3_SPDIF 0x0092 |
13017 | 35 #define WAVE_FORMAT_EXTENSIBLE 0xFFFE |
36 | |
37 static const GUID KSDATAFORMAT_SUBTYPE_PCM = { | |
38 0x1,0x0000,0x0010,{0x80,0x00,0x00,0xaa,0x00,0x38,0x9b,0x71} | |
39 }; | |
40 | |
41 typedef struct { | |
42 WAVEFORMATEX Format; | |
43 union { | |
44 WORD wValidBitsPerSample; | |
45 WORD wSamplesPerBlock; | |
46 WORD wReserved; | |
47 } Samples; | |
48 DWORD dwChannelMask; | |
49 GUID SubFormat; | |
50 } WAVEFORMATEXTENSIBLE, *PWAVEFORMATEXTENSIBLE; | |
51 | |
52 #define SPEAKER_FRONT_LEFT 0x1 | |
53 #define SPEAKER_FRONT_RIGHT 0x2 | |
54 #define SPEAKER_FRONT_CENTER 0x4 | |
55 #define SPEAKER_LOW_FREQUENCY 0x8 | |
56 #define SPEAKER_BACK_LEFT 0x10 | |
57 #define SPEAKER_BACK_RIGHT 0x20 | |
58 #define SPEAKER_FRONT_LEFT_OF_CENTER 0x40 | |
59 #define SPEAKER_FRONT_RIGHT_OF_CENTER 0x80 | |
60 #define SPEAKER_BACK_CENTER 0x100 | |
61 #define SPEAKER_SIDE_LEFT 0x200 | |
62 #define SPEAKER_SIDE_RIGHT 0x400 | |
63 #define SPEAKER_TOP_CENTER 0x800 | |
64 #define SPEAKER_TOP_FRONT_LEFT 0x1000 | |
65 #define SPEAKER_TOP_FRONT_CENTER 0x2000 | |
66 #define SPEAKER_TOP_FRONT_RIGHT 0x4000 | |
67 #define SPEAKER_TOP_BACK_LEFT 0x8000 | |
68 #define SPEAKER_TOP_BACK_CENTER 0x10000 | |
69 #define SPEAKER_TOP_BACK_RIGHT 0x20000 | |
70 | |
71 static const int channel_mask[] = { | |
72 SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_LOW_FREQUENCY, | |
73 SPEAKER_FRONT_LEFT | SPEAKER_FRONT_CENTER | SPEAKER_FRONT_RIGHT | SPEAKER_LOW_FREQUENCY, | |
74 SPEAKER_FRONT_LEFT | SPEAKER_FRONT_CENTER | SPEAKER_FRONT_RIGHT | SPEAKER_BACK_CENTER | SPEAKER_LOW_FREQUENCY, | |
75 SPEAKER_FRONT_LEFT | SPEAKER_FRONT_CENTER | SPEAKER_FRONT_RIGHT | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | SPEAKER_LOW_FREQUENCY | |
76 }; | |
77 | |
78 | |
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79 |
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80 #define SAMPLESIZE 1024 |
7913 | 81 #define BUFFER_SIZE 4096 |
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82 #define BUFFER_COUNT 16 |
7913 | 83 |
84 | |
85 static WAVEHDR* waveBlocks; //pointer to our ringbuffer memory | |
86 static HWAVEOUT hWaveOut; //handle to the waveout device | |
87 static unsigned int buf_write=0; | |
88 static unsigned int buf_write_pos=0; | |
89 static int full_buffers=0; | |
90 static int buffered_bytes=0; | |
91 | |
92 | |
93 static ao_info_t info = | |
94 { | |
95 "Windows waveOut audio output", | |
96 "win32", | |
97 "Sascha Sommer <saschasommer@freenet.de>", | |
98 "" | |
99 }; | |
100 | |
101 LIBAO_EXTERN(win32) | |
102 | |
103 static void CALLBACK waveOutProc(HWAVEOUT hWaveOut,UINT uMsg,DWORD dwInstance, | |
104 DWORD dwParam1,DWORD dwParam2) | |
105 { | |
106 if(uMsg != WOM_DONE) | |
107 return; | |
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108 if (full_buffers) { |
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109 buffered_bytes-=BUFFER_SIZE; |
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110 --full_buffers; |
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111 } else { |
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112 buffered_bytes=0; |
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113 } |
7913 | 114 } |
115 | |
116 // to set/get/query special features/parameters | |
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117 static int control(int cmd,void *arg) |
7913 | 118 { |
119 DWORD volume; | |
120 switch (cmd) | |
121 { | |
122 case AOCONTROL_GET_VOLUME: | |
123 { | |
124 ao_control_vol_t* vol = (ao_control_vol_t*)arg; | |
125 waveOutGetVolume(hWaveOut,&volume); | |
126 vol->left = (float)(LOWORD(volume)/655.35); | |
127 vol->right = (float)(HIWORD(volume)/655.35); | |
128 mp_msg(MSGT_AO, MSGL_DBG2,"ao_win32: volume left:%f volume right:%f\n",vol->left,vol->right); | |
129 return CONTROL_OK; | |
130 } | |
131 case AOCONTROL_SET_VOLUME: | |
132 { | |
133 ao_control_vol_t* vol = (ao_control_vol_t*)arg; | |
134 volume = MAKELONG(vol->left*655.35,vol->right*655.35); | |
135 waveOutSetVolume(hWaveOut,volume); | |
136 return CONTROL_OK; | |
137 } | |
138 } | |
139 return -1; | |
140 } | |
141 | |
142 // open & setup audio device | |
143 // return: 1=success 0=fail | |
144 static int init(int rate,int channels,int format,int flags) | |
145 { | |
13017 | 146 WAVEFORMATEXTENSIBLE wformat; |
7913 | 147 DWORD totalBufferSize = (BUFFER_SIZE + sizeof(WAVEHDR)) * BUFFER_COUNT; |
148 MMRESULT result; | |
149 unsigned char* buffer; | |
150 int i; | |
151 | |
14135 | 152 switch(format){ |
14245 | 153 case AF_FORMAT_AC3: |
154 case AF_FORMAT_S24_LE: | |
155 case AF_FORMAT_S16_LE: | |
156 case AF_FORMAT_S8: | |
14135 | 157 break; |
158 default: | |
14264 | 159 mp_msg(MSGT_AO, MSGL_V,"ao_win32: format %s not supported defaulting to Signed 16-bit Little-Endian\n",af_fmt2str_short(format)); |
14245 | 160 format=AF_FORMAT_S16_LE; |
14135 | 161 } |
7913 | 162 //fill global ao_data |
163 ao_data.channels=channels; | |
164 ao_data.samplerate=rate; | |
165 ao_data.format=format; | |
166 ao_data.bps=channels*rate; | |
14245 | 167 if(format != AF_FORMAT_U8 && format != AF_FORMAT_S8) |
7913 | 168 ao_data.bps*=2; |
169 if(ao_data.buffersize==-1) | |
170 { | |
14264 | 171 ao_data.buffersize=af_fmt2bits(format)/8; |
7913 | 172 ao_data.buffersize*= channels; |
173 ao_data.buffersize*= SAMPLESIZE; | |
174 } | |
14264 | 175 mp_msg(MSGT_AO, MSGL_V,"ao_win32: Samplerate:%iHz Channels:%i Format:%s\n",rate, channels, af_fmt2str_short(format)); |
7913 | 176 mp_msg(MSGT_AO, MSGL_V,"ao_win32: Buffersize:%d\n",ao_data.buffersize); |
177 | |
178 //fill waveformatex | |
13017 | 179 ZeroMemory( &wformat, sizeof(WAVEFORMATEXTENSIBLE)); |
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180 wformat.Format.cbSize = (channels>2)?sizeof(WAVEFORMATEXTENSIBLE)-sizeof(WAVEFORMATEX):0; |
13017 | 181 wformat.Format.nChannels = channels; |
182 wformat.Format.nSamplesPerSec = rate; | |
14245 | 183 if(format == AF_FORMAT_AC3) |
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184 { |
13017 | 185 wformat.Format.wFormatTag = WAVE_FORMAT_DOLBY_AC3_SPDIF; |
186 wformat.Format.wBitsPerSample = 16; | |
187 wformat.Format.nBlockAlign = 4; | |
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188 } |
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189 else |
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190 { |
13017 | 191 wformat.Format.wFormatTag = (channels>2)?WAVE_FORMAT_EXTENSIBLE:WAVE_FORMAT_PCM; |
14264 | 192 wformat.Format.wBitsPerSample = af_fmt2bits(format); |
13017 | 193 wformat.Format.nBlockAlign = wformat.Format.nChannels * (wformat.Format.wBitsPerSample >> 3); |
194 } | |
195 if(channels>2) | |
196 { | |
197 wformat.dwChannelMask = channel_mask[channels-3]; | |
198 wformat.SubFormat = KSDATAFORMAT_SUBTYPE_PCM; | |
14264 | 199 wformat.Samples.wValidBitsPerSample=af_fmt2bits(format); |
13017 | 200 } |
201 | |
202 wformat.Format.nAvgBytesPerSec = wformat.Format.nSamplesPerSec * wformat.Format.nBlockAlign; | |
7913 | 203 |
204 //open sound device | |
205 //WAVE_MAPPER always points to the default wave device on the system | |
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206 result = waveOutOpen(&hWaveOut,WAVE_MAPPER,(WAVEFORMATEX*)&wformat,(DWORD_PTR)waveOutProc,0,CALLBACK_FUNCTION); |
7913 | 207 if(result == WAVERR_BADFORMAT) |
208 { | |
209 mp_msg(MSGT_AO, MSGL_ERR,"ao_win32: format not supported switching to default\n"); | |
13017 | 210 ao_data.channels = wformat.Format.nChannels = 2; |
211 ao_data.samplerate = wformat.Format.nSamplesPerSec = 44100; | |
14245 | 212 ao_data.format = AF_FORMAT_S16_LE; |
13017 | 213 ao_data.bps=ao_data.channels * ao_data.samplerate*2; |
214 wformat.Format.wBitsPerSample=16; | |
215 wformat.Format.wFormatTag=WAVE_FORMAT_PCM; | |
216 wformat.Format.nBlockAlign = wformat.Format.nChannels * (wformat.Format.wBitsPerSample >> 3); | |
217 wformat.Format.nAvgBytesPerSec = wformat.Format.nSamplesPerSec * wformat.Format.nBlockAlign; | |
218 ao_data.buffersize=(wformat.Format.wBitsPerSample>>3)*wformat.Format.nChannels*SAMPLESIZE; | |
219 result = waveOutOpen(&hWaveOut,WAVE_MAPPER,(WAVEFORMATEX*)&wformat,(DWORD_PTR)waveOutProc,0,CALLBACK_FUNCTION); | |
7913 | 220 } |
221 if(result != MMSYSERR_NOERROR) | |
222 { | |
14503 | 223 mp_msg(MSGT_AO, MSGL_ERR,"ao_win32: unable to open wave mapper device (result=%i)\n",result); |
7913 | 224 return 0; |
225 } | |
226 //allocate buffer memory as one big block | |
227 buffer = malloc(totalBufferSize); | |
9589 | 228 memset(buffer,0x0,totalBufferSize); |
7913 | 229 //and setup pointers to each buffer |
230 waveBlocks = (WAVEHDR*)buffer; | |
231 buffer += sizeof(WAVEHDR) * BUFFER_COUNT; | |
232 for(i = 0; i < BUFFER_COUNT; i++) { | |
233 waveBlocks[i].lpData = buffer; | |
234 buffer += BUFFER_SIZE; | |
235 } | |
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236 buf_write=0; |
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237 buf_write_pos=0; |
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238 full_buffers=0; |
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239 buffered_bytes=0; |
7913 | 240 |
241 return 1; | |
242 } | |
243 | |
244 // close audio device | |
12145 | 245 static void uninit(int immed) |
7913 | 246 { |
12146 | 247 if(!immed)while(buffered_bytes > 0)usec_sleep(50000); |
248 else buffered_bytes=0; | |
7913 | 249 waveOutReset(hWaveOut); |
250 waveOutClose(hWaveOut); | |
251 mp_msg(MSGT_AO, MSGL_V,"waveOut device closed\n"); | |
252 free(waveBlocks); | |
253 mp_msg(MSGT_AO, MSGL_V,"buffer memory freed\n"); | |
254 } | |
255 | |
256 // stop playing and empty buffers (for seeking/pause) | |
257 static void reset() | |
258 { | |
259 waveOutReset(hWaveOut); | |
260 buf_write=0; | |
261 buf_write_pos=0; | |
262 full_buffers=0; | |
263 buffered_bytes=0; | |
264 } | |
265 | |
266 // stop playing, keep buffers (for pause) | |
267 static void audio_pause() | |
268 { | |
269 waveOutPause(hWaveOut); | |
270 } | |
271 | |
272 // resume playing, after audio_pause() | |
273 static void audio_resume() | |
274 { | |
275 waveOutRestart(hWaveOut); | |
276 } | |
277 | |
278 // return: how many bytes can be played without blocking | |
279 static int get_space() | |
280 { | |
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281 return BUFFER_COUNT*BUFFER_SIZE - buffered_bytes; |
7913 | 282 } |
283 | |
284 //writes data into buffer, based on ringbuffer code in ao_sdl.c | |
285 static int write_waveOutBuffer(unsigned char* data,int len){ | |
286 WAVEHDR* current; | |
287 int len2=0; | |
288 int x; | |
289 while(len>0){ | |
290 current = &waveBlocks[buf_write]; | |
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291 if(buffered_bytes==BUFFER_COUNT*BUFFER_SIZE) break; |
7913 | 292 //unprepare the header if it is prepared |
293 if(current->dwFlags & WHDR_PREPARED) | |
294 waveOutUnprepareHeader(hWaveOut, current, sizeof(WAVEHDR)); | |
295 x=BUFFER_SIZE-buf_write_pos; | |
296 if(x>len) x=len; | |
297 memcpy(current->lpData+buf_write_pos,data+len2,x); | |
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298 if(buf_write_pos==0)full_buffers++; |
7913 | 299 len2+=x; len-=x; |
300 buffered_bytes+=x; buf_write_pos+=x; | |
301 //prepare header and write data to device | |
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302 current->dwBufferLength = buf_write_pos; |
7913 | 303 waveOutPrepareHeader(hWaveOut, current, sizeof(WAVEHDR)); |
304 waveOutWrite(hWaveOut, current, sizeof(WAVEHDR)); | |
305 | |
306 if(buf_write_pos>=BUFFER_SIZE){ //buffer is full find next | |
307 // block is full, find next! | |
308 buf_write=(buf_write+1)%BUFFER_COUNT; | |
309 buf_write_pos=0; | |
310 } | |
311 } | |
312 return len2; | |
313 } | |
314 | |
315 // plays 'len' bytes of 'data' | |
316 // it should round it down to outburst*n | |
317 // return: number of bytes played | |
318 static int play(void* data,int len,int flags) | |
319 { | |
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320 len = (len/ao_data.outburst)*ao_data.outburst; |
7913 | 321 return write_waveOutBuffer(data,len); |
322 } | |
10106 | 323 |
7913 | 324 // return: delay in seconds between first and last sample in buffer |
325 static float get_delay() | |
326 { | |
327 return (float)(buffered_bytes + ao_data.buffersize)/(float)ao_data.bps; | |
328 } |