Add support for VDPAU video out, including hardware decoding.
[mplayer/glamo.git] / libao2 / ao_win32.c
blob40830eb23353dbc466b5e84c3c6f133b99741f91
1 /*
2 * Windows waveOut interface
4 * Copyright (c) 2002 - 2004 Sascha Sommer <saschasommer@freenet.de>
6 * This file is part of MPlayer.
8 * MPlayer is free software; you can redistribute it and/or modify
9 * it under the terms of the GNU General Public License as published by
10 * the Free Software Foundation; either version 2 of the License, or
11 * (at your option) any later version.
13 * MPlayer is distributed in the hope that it will be useful,
14 * but WITHOUT ANY WARRANTY; without even the implied warranty of
15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
16 * GNU General Public License for more details.
18 * You should have received a copy of the GNU General Public License along
19 * with MPlayer; if not, write to the Free Software Foundation, Inc.,
20 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
23 #include <stdio.h>
24 #include <stdlib.h>
25 #include <windows.h>
26 #include <mmsystem.h>
28 #include "config.h"
29 #include "libaf/af_format.h"
30 #include "audio_out.h"
31 #include "audio_out_internal.h"
32 #include "mp_msg.h"
33 #include "libvo/fastmemcpy.h"
34 #include "osdep/timer.h"
36 #define WAVE_FORMAT_DOLBY_AC3_SPDIF 0x0092
37 #define WAVE_FORMAT_EXTENSIBLE 0xFFFE
39 static const GUID KSDATAFORMAT_SUBTYPE_PCM = {
40 0x1,0x0000,0x0010,{0x80,0x00,0x00,0xaa,0x00,0x38,0x9b,0x71}
43 typedef struct {
44 WAVEFORMATEX Format;
45 union {
46 WORD wValidBitsPerSample;
47 WORD wSamplesPerBlock;
48 WORD wReserved;
49 } Samples;
50 DWORD dwChannelMask;
51 GUID SubFormat;
52 } WAVEFORMATEXTENSIBLE, *PWAVEFORMATEXTENSIBLE;
54 #define SPEAKER_FRONT_LEFT 0x1
55 #define SPEAKER_FRONT_RIGHT 0x2
56 #define SPEAKER_FRONT_CENTER 0x4
57 #define SPEAKER_LOW_FREQUENCY 0x8
58 #define SPEAKER_BACK_LEFT 0x10
59 #define SPEAKER_BACK_RIGHT 0x20
60 #define SPEAKER_FRONT_LEFT_OF_CENTER 0x40
61 #define SPEAKER_FRONT_RIGHT_OF_CENTER 0x80
62 #define SPEAKER_BACK_CENTER 0x100
63 #define SPEAKER_SIDE_LEFT 0x200
64 #define SPEAKER_SIDE_RIGHT 0x400
65 #define SPEAKER_TOP_CENTER 0x800
66 #define SPEAKER_TOP_FRONT_LEFT 0x1000
67 #define SPEAKER_TOP_FRONT_CENTER 0x2000
68 #define SPEAKER_TOP_FRONT_RIGHT 0x4000
69 #define SPEAKER_TOP_BACK_LEFT 0x8000
70 #define SPEAKER_TOP_BACK_CENTER 0x10000
71 #define SPEAKER_TOP_BACK_RIGHT 0x20000
73 static const int channel_mask[] = {
74 SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_LOW_FREQUENCY,
75 SPEAKER_FRONT_LEFT | SPEAKER_FRONT_CENTER | SPEAKER_FRONT_RIGHT | SPEAKER_LOW_FREQUENCY,
76 SPEAKER_FRONT_LEFT | SPEAKER_FRONT_CENTER | SPEAKER_FRONT_RIGHT | SPEAKER_BACK_CENTER | SPEAKER_LOW_FREQUENCY,
77 SPEAKER_FRONT_LEFT | SPEAKER_FRONT_CENTER | SPEAKER_FRONT_RIGHT | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | SPEAKER_LOW_FREQUENCY
82 #define SAMPLESIZE 1024
83 #define BUFFER_SIZE 4096
84 #define BUFFER_COUNT 16
87 static WAVEHDR* waveBlocks; //pointer to our ringbuffer memory
88 static HWAVEOUT hWaveOut; //handle to the waveout device
89 static unsigned int buf_write=0;
90 static unsigned int buf_write_pos=0;
91 static int full_buffers=0;
92 static int buffered_bytes=0;
95 static ao_info_t info =
97 "Windows waveOut audio output",
98 "win32",
99 "Sascha Sommer <saschasommer@freenet.de>",
103 LIBAO_EXTERN(win32)
105 static void CALLBACK waveOutProc(HWAVEOUT hWaveOut,UINT uMsg,DWORD dwInstance,
106 DWORD dwParam1,DWORD dwParam2)
108 if(uMsg != WOM_DONE)
109 return;
110 if (full_buffers) {
111 buffered_bytes-=BUFFER_SIZE;
112 --full_buffers;
113 } else {
114 buffered_bytes=0;
118 // to set/get/query special features/parameters
119 static int control(int cmd,void *arg)
121 DWORD volume;
122 switch (cmd)
124 case AOCONTROL_GET_VOLUME:
126 ao_control_vol_t* vol = (ao_control_vol_t*)arg;
127 waveOutGetVolume(hWaveOut,&volume);
128 vol->left = (float)(LOWORD(volume)/655.35);
129 vol->right = (float)(HIWORD(volume)/655.35);
130 mp_msg(MSGT_AO, MSGL_DBG2,"ao_win32: volume left:%f volume right:%f\n",vol->left,vol->right);
131 return CONTROL_OK;
133 case AOCONTROL_SET_VOLUME:
135 ao_control_vol_t* vol = (ao_control_vol_t*)arg;
136 volume = MAKELONG(vol->left*655.35,vol->right*655.35);
137 waveOutSetVolume(hWaveOut,volume);
138 return CONTROL_OK;
141 return -1;
144 // open & setup audio device
145 // return: 1=success 0=fail
146 static int init(int rate,int channels,int format,int flags)
148 WAVEFORMATEXTENSIBLE wformat;
149 DWORD totalBufferSize = (BUFFER_SIZE + sizeof(WAVEHDR)) * BUFFER_COUNT;
150 MMRESULT result;
151 unsigned char* buffer;
152 int i;
154 switch(format){
155 case AF_FORMAT_AC3:
156 case AF_FORMAT_S24_LE:
157 case AF_FORMAT_S16_LE:
158 case AF_FORMAT_S8:
159 break;
160 default:
161 mp_msg(MSGT_AO, MSGL_V,"ao_win32: format %s not supported defaulting to Signed 16-bit Little-Endian\n",af_fmt2str_short(format));
162 format=AF_FORMAT_S16_LE;
165 // FIXME multichannel mode is buggy
166 if(channels > 2)
167 channels = 2;
169 //fill global ao_data
170 ao_data.channels=channels;
171 ao_data.samplerate=rate;
172 ao_data.format=format;
173 ao_data.bps=channels*rate;
174 if(format != AF_FORMAT_U8 && format != AF_FORMAT_S8)
175 ao_data.bps*=2;
176 if(ao_data.buffersize==-1)
178 ao_data.buffersize=af_fmt2bits(format)/8;
179 ao_data.buffersize*= channels;
180 ao_data.buffersize*= SAMPLESIZE;
182 mp_msg(MSGT_AO, MSGL_V,"ao_win32: Samplerate:%iHz Channels:%i Format:%s\n",rate, channels, af_fmt2str_short(format));
183 mp_msg(MSGT_AO, MSGL_V,"ao_win32: Buffersize:%d\n",ao_data.buffersize);
185 //fill waveformatex
186 ZeroMemory( &wformat, sizeof(WAVEFORMATEXTENSIBLE));
187 wformat.Format.cbSize = (channels>2)?sizeof(WAVEFORMATEXTENSIBLE)-sizeof(WAVEFORMATEX):0;
188 wformat.Format.nChannels = channels;
189 wformat.Format.nSamplesPerSec = rate;
190 if(format == AF_FORMAT_AC3)
192 wformat.Format.wFormatTag = WAVE_FORMAT_DOLBY_AC3_SPDIF;
193 wformat.Format.wBitsPerSample = 16;
194 wformat.Format.nBlockAlign = 4;
196 else
198 wformat.Format.wFormatTag = (channels>2)?WAVE_FORMAT_EXTENSIBLE:WAVE_FORMAT_PCM;
199 wformat.Format.wBitsPerSample = af_fmt2bits(format);
200 wformat.Format.nBlockAlign = wformat.Format.nChannels * (wformat.Format.wBitsPerSample >> 3);
202 if(channels>2)
204 wformat.dwChannelMask = channel_mask[channels-3];
205 wformat.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
206 wformat.Samples.wValidBitsPerSample=af_fmt2bits(format);
209 wformat.Format.nAvgBytesPerSec = wformat.Format.nSamplesPerSec * wformat.Format.nBlockAlign;
211 //open sound device
212 //WAVE_MAPPER always points to the default wave device on the system
213 result = waveOutOpen(&hWaveOut,WAVE_MAPPER,(WAVEFORMATEX*)&wformat,(DWORD_PTR)waveOutProc,0,CALLBACK_FUNCTION);
214 if(result == WAVERR_BADFORMAT)
216 mp_msg(MSGT_AO, MSGL_ERR,"ao_win32: format not supported switching to default\n");
217 ao_data.channels = wformat.Format.nChannels = 2;
218 ao_data.samplerate = wformat.Format.nSamplesPerSec = 44100;
219 ao_data.format = AF_FORMAT_S16_LE;
220 ao_data.bps=ao_data.channels * ao_data.samplerate*2;
221 wformat.Format.wBitsPerSample=16;
222 wformat.Format.wFormatTag=WAVE_FORMAT_PCM;
223 wformat.Format.nBlockAlign = wformat.Format.nChannels * (wformat.Format.wBitsPerSample >> 3);
224 wformat.Format.nAvgBytesPerSec = wformat.Format.nSamplesPerSec * wformat.Format.nBlockAlign;
225 ao_data.buffersize=(wformat.Format.wBitsPerSample>>3)*wformat.Format.nChannels*SAMPLESIZE;
226 result = waveOutOpen(&hWaveOut,WAVE_MAPPER,(WAVEFORMATEX*)&wformat,(DWORD_PTR)waveOutProc,0,CALLBACK_FUNCTION);
228 if(result != MMSYSERR_NOERROR)
230 mp_msg(MSGT_AO, MSGL_ERR,"ao_win32: unable to open wave mapper device (result=%i)\n",result);
231 return 0;
233 //allocate buffer memory as one big block
234 buffer = malloc(totalBufferSize);
235 memset(buffer,0x0,totalBufferSize);
236 //and setup pointers to each buffer
237 waveBlocks = (WAVEHDR*)buffer;
238 buffer += sizeof(WAVEHDR) * BUFFER_COUNT;
239 for(i = 0; i < BUFFER_COUNT; i++) {
240 waveBlocks[i].lpData = buffer;
241 buffer += BUFFER_SIZE;
243 buf_write=0;
244 buf_write_pos=0;
245 full_buffers=0;
246 buffered_bytes=0;
248 return 1;
251 // close audio device
252 static void uninit(int immed)
254 if(!immed)while(buffered_bytes > 0)usec_sleep(50000);
255 else buffered_bytes=0;
256 waveOutReset(hWaveOut);
257 waveOutClose(hWaveOut);
258 mp_msg(MSGT_AO, MSGL_V,"waveOut device closed\n");
259 free(waveBlocks);
260 mp_msg(MSGT_AO, MSGL_V,"buffer memory freed\n");
263 // stop playing and empty buffers (for seeking/pause)
264 static void reset(void)
266 waveOutReset(hWaveOut);
267 buf_write=0;
268 buf_write_pos=0;
269 full_buffers=0;
270 buffered_bytes=0;
273 // stop playing, keep buffers (for pause)
274 static void audio_pause(void)
276 waveOutPause(hWaveOut);
279 // resume playing, after audio_pause()
280 static void audio_resume(void)
282 waveOutRestart(hWaveOut);
285 // return: how many bytes can be played without blocking
286 static int get_space(void)
288 return BUFFER_COUNT*BUFFER_SIZE - buffered_bytes;
291 //writes data into buffer, based on ringbuffer code in ao_sdl.c
292 static int write_waveOutBuffer(unsigned char* data,int len){
293 WAVEHDR* current;
294 int len2=0;
295 int x;
296 while(len>0){
297 current = &waveBlocks[buf_write];
298 if(buffered_bytes==BUFFER_COUNT*BUFFER_SIZE) break;
299 //unprepare the header if it is prepared
300 if(current->dwFlags & WHDR_PREPARED)
301 waveOutUnprepareHeader(hWaveOut, current, sizeof(WAVEHDR));
302 x=BUFFER_SIZE-buf_write_pos;
303 if(x>len) x=len;
304 fast_memcpy(current->lpData+buf_write_pos,data+len2,x);
305 if(buf_write_pos==0)full_buffers++;
306 len2+=x; len-=x;
307 buffered_bytes+=x; buf_write_pos+=x;
308 //prepare header and write data to device
309 current->dwBufferLength = buf_write_pos;
310 waveOutPrepareHeader(hWaveOut, current, sizeof(WAVEHDR));
311 waveOutWrite(hWaveOut, current, sizeof(WAVEHDR));
313 if(buf_write_pos>=BUFFER_SIZE){ //buffer is full find next
314 // block is full, find next!
315 buf_write=(buf_write+1)%BUFFER_COUNT;
316 buf_write_pos=0;
319 return len2;
322 // plays 'len' bytes of 'data'
323 // it should round it down to outburst*n
324 // return: number of bytes played
325 static int play(void* data,int len,int flags)
327 if (!(flags & AOPLAY_FINAL_CHUNK))
328 len = (len/ao_data.outburst)*ao_data.outburst;
329 return write_waveOutBuffer(data,len);
332 // return: delay in seconds between first and last sample in buffer
333 static float get_delay(void)
335 return (float)(buffered_bytes + ao_data.buffersize)/(float)ao_data.bps;