aboutsummaryrefslogtreecommitdiffhomepage
path: root/libao2/ao_win32.c
blob: 55ed17b457f475aedc4903dadaac5684d98b945d (plain)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
/*
 * Windows waveOut interface
 *
 * Copyright (c) 2002 - 2004 Sascha Sommer <saschasommer@freenet.de>
 *
 * This file is part of MPlayer.
 *
 * MPlayer is free software; you can redistribute it and/or modify
 * it under the terms of the GNU General Public License as published by
 * the Free Software Foundation; either version 2 of the License, or
 * (at your option) any later version.
 *
 * MPlayer is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 * GNU General Public License for more details.
 *
 * You should have received a copy of the GNU General Public License along
 * with MPlayer; if not, write to the Free Software Foundation, Inc.,
 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
 */

#include <stdio.h>
#include <stdlib.h>
#include <windows.h>
#include <mmsystem.h>

#include "config.h"
#include "libaf/af_format.h"
#include "audio_out.h"
#include "audio_out_internal.h"
#include "mp_msg.h"
#include "libvo/fastmemcpy.h"
#include "osdep/timer.h"

#define WAVE_FORMAT_DOLBY_AC3_SPDIF 0x0092
#define WAVE_FORMAT_EXTENSIBLE      0xFFFE

static const  GUID KSDATAFORMAT_SUBTYPE_PCM = {
	0x1,0x0000,0x0010,{0x80,0x00,0x00,0xaa,0x00,0x38,0x9b,0x71}
};

typedef struct {
  WAVEFORMATEX  Format;
  union {
    WORD  wValidBitsPerSample;
    WORD  wSamplesPerBlock;
    WORD  wReserved;
  } Samples;
  DWORD  dwChannelMask;
  GUID  SubFormat;
} WAVEFORMATEXTENSIBLE, *PWAVEFORMATEXTENSIBLE;

#define SPEAKER_FRONT_LEFT              0x1
#define SPEAKER_FRONT_RIGHT             0x2
#define SPEAKER_FRONT_CENTER            0x4
#define SPEAKER_LOW_FREQUENCY           0x8
#define SPEAKER_BACK_LEFT               0x10
#define SPEAKER_BACK_RIGHT              0x20
#define SPEAKER_FRONT_LEFT_OF_CENTER    0x40
#define SPEAKER_FRONT_RIGHT_OF_CENTER   0x80
#define SPEAKER_BACK_CENTER             0x100
#define SPEAKER_SIDE_LEFT               0x200
#define SPEAKER_SIDE_RIGHT              0x400
#define SPEAKER_TOP_CENTER              0x800
#define SPEAKER_TOP_FRONT_LEFT          0x1000
#define SPEAKER_TOP_FRONT_CENTER        0x2000
#define SPEAKER_TOP_FRONT_RIGHT         0x4000
#define SPEAKER_TOP_BACK_LEFT           0x8000
#define SPEAKER_TOP_BACK_CENTER         0x10000
#define SPEAKER_TOP_BACK_RIGHT          0x20000

static const int channel_mask[] = {
  SPEAKER_FRONT_LEFT   | SPEAKER_FRONT_RIGHT  | SPEAKER_LOW_FREQUENCY,
  SPEAKER_FRONT_LEFT   | SPEAKER_FRONT_CENTER | SPEAKER_FRONT_RIGHT  | SPEAKER_LOW_FREQUENCY,
  SPEAKER_FRONT_LEFT   | SPEAKER_FRONT_CENTER | SPEAKER_FRONT_RIGHT  | SPEAKER_BACK_CENTER  | SPEAKER_LOW_FREQUENCY,
  SPEAKER_FRONT_LEFT   | SPEAKER_FRONT_CENTER | SPEAKER_FRONT_RIGHT  | SPEAKER_BACK_LEFT    | SPEAKER_BACK_RIGHT     | SPEAKER_LOW_FREQUENCY
};



#define SAMPLESIZE   1024
#define BUFFER_SIZE  4096
#define BUFFER_COUNT 16


static WAVEHDR*     waveBlocks;         //pointer to our ringbuffer memory
static HWAVEOUT     hWaveOut;           //handle to the waveout device
static unsigned int buf_write=0;
static volatile int buf_read=0;


static const ao_info_t info =
{
	"Windows waveOut audio output",
	"win32",
	"Sascha Sommer <saschasommer@freenet.de>",
	""
};

LIBAO_EXTERN(win32)

static void CALLBACK waveOutProc(HWAVEOUT hWaveOut,UINT uMsg,DWORD dwInstance,
    DWORD dwParam1,DWORD dwParam2)
{
	if(uMsg != WOM_DONE)
        return;
	buf_read = (buf_read + 1) % BUFFER_COUNT;
}

// to set/get/query special features/parameters
static int control(int cmd,void *arg)
{
	DWORD volume;
	switch (cmd)
	{
		case AOCONTROL_GET_VOLUME:
		{
			ao_control_vol_t* vol = (ao_control_vol_t*)arg;
			waveOutGetVolume(hWaveOut,&volume);
			vol->left = (float)(LOWORD(volume)/655.35);
			vol->right = (float)(HIWORD(volume)/655.35);
			mp_msg(MSGT_AO, MSGL_DBG2,"ao_win32: volume left:%f volume right:%f\n",vol->left,vol->right);
			return CONTROL_OK;
		}
		case AOCONTROL_SET_VOLUME:
		{
			ao_control_vol_t* vol = (ao_control_vol_t*)arg;
			volume = MAKELONG(vol->left*655.35,vol->right*655.35);
			waveOutSetVolume(hWaveOut,volume);
			return CONTROL_OK;
		}
	}
    return -1;
}

// open & setup audio device
// return: 1=success 0=fail
static int init(int rate,int channels,int format,int flags)
{
	WAVEFORMATEXTENSIBLE wformat;
	MMRESULT result;
	unsigned char* buffer;
	int i;

	if (AF_FORMAT_IS_AC3(format))
		format = AF_FORMAT_AC3_NE;
	switch(format){
		case AF_FORMAT_AC3_NE:
		case AF_FORMAT_S24_LE:
		case AF_FORMAT_S16_LE:
		case AF_FORMAT_U8:
			break;
		default:
			mp_msg(MSGT_AO, MSGL_V,"ao_win32: format %s not supported defaulting to Signed 16-bit Little-Endian\n",af_fmt2str_short(format));
			format=AF_FORMAT_S16_LE;
	}

	// FIXME multichannel mode is buggy
	if(channels > 2)
		channels = 2;

	//fill global ao_data
	ao_data.channels=channels;
	ao_data.samplerate=rate;
	ao_data.format=format;
	ao_data.bps=channels*rate;
	if(format != AF_FORMAT_U8 && format != AF_FORMAT_S8)
	  ao_data.bps*=2;
	ao_data.outburst = BUFFER_SIZE;
	if(ao_data.buffersize==-1)
	{
		ao_data.buffersize=af_fmt2bits(format)/8;
        ao_data.buffersize*= channels;
		ao_data.buffersize*= SAMPLESIZE;
	}
	mp_msg(MSGT_AO, MSGL_V,"ao_win32: Samplerate:%iHz Channels:%i Format:%s\n",rate, channels, af_fmt2str_short(format));
    mp_msg(MSGT_AO, MSGL_V,"ao_win32: Buffersize:%d\n",ao_data.buffersize);

	//fill waveformatex
    ZeroMemory( &wformat, sizeof(WAVEFORMATEXTENSIBLE));
    wformat.Format.cbSize          = (channels>2)?sizeof(WAVEFORMATEXTENSIBLE)-sizeof(WAVEFORMATEX):0;
    wformat.Format.nChannels       = channels;
    wformat.Format.nSamplesPerSec  = rate;
    if(AF_FORMAT_IS_AC3(format))
    {
        wformat.Format.wFormatTag      = WAVE_FORMAT_DOLBY_AC3_SPDIF;
        wformat.Format.wBitsPerSample  = 16;
        wformat.Format.nBlockAlign     = 4;
    }
    else
    {
        wformat.Format.wFormatTag      = (channels>2)?WAVE_FORMAT_EXTENSIBLE:WAVE_FORMAT_PCM;
        wformat.Format.wBitsPerSample  = af_fmt2bits(format);
        wformat.Format.nBlockAlign     = wformat.Format.nChannels * (wformat.Format.wBitsPerSample >> 3);
    }
	if(channels>2)
	{
        wformat.dwChannelMask = channel_mask[channels-3];
        wformat.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
	    wformat.Samples.wValidBitsPerSample=af_fmt2bits(format);
    }

    wformat.Format.nAvgBytesPerSec = wformat.Format.nSamplesPerSec * wformat.Format.nBlockAlign;

    //open sound device
    //WAVE_MAPPER always points to the default wave device on the system
    result = waveOutOpen(&hWaveOut,WAVE_MAPPER,(WAVEFORMATEX*)&wformat,(DWORD_PTR)waveOutProc,0,CALLBACK_FUNCTION);
	if(result == WAVERR_BADFORMAT)
	{
		mp_msg(MSGT_AO, MSGL_ERR,"ao_win32: format not supported switching to default\n");
        ao_data.channels = wformat.Format.nChannels = 2;
	    ao_data.samplerate = wformat.Format.nSamplesPerSec = 44100;
	    ao_data.format = AF_FORMAT_S16_LE;
		ao_data.bps=ao_data.channels * ao_data.samplerate*2;
	    wformat.Format.wBitsPerSample=16;
        wformat.Format.wFormatTag=WAVE_FORMAT_PCM;
		wformat.Format.nBlockAlign     = wformat.Format.nChannels * (wformat.Format.wBitsPerSample >> 3);
        wformat.Format.nAvgBytesPerSec = wformat.Format.nSamplesPerSec * wformat.Format.nBlockAlign;
		ao_data.buffersize=(wformat.Format.wBitsPerSample>>3)*wformat.Format.nChannels*SAMPLESIZE;
        result = waveOutOpen(&hWaveOut,WAVE_MAPPER,(WAVEFORMATEX*)&wformat,(DWORD_PTR)waveOutProc,0,CALLBACK_FUNCTION);
	}
	if(result != MMSYSERR_NOERROR)
	{
		mp_msg(MSGT_AO, MSGL_ERR,"ao_win32: unable to open wave mapper device (result=%i)\n",result);
		return 0;
    }
	//allocate buffer memory as one big block
	buffer = calloc(BUFFER_COUNT, BUFFER_SIZE + sizeof(WAVEHDR));
    //and setup pointers to each buffer
    waveBlocks = (WAVEHDR*)buffer;
    buffer += sizeof(WAVEHDR) * BUFFER_COUNT;
    for(i = 0; i < BUFFER_COUNT; i++) {
        waveBlocks[i].lpData = buffer;
        buffer += BUFFER_SIZE;
    }
    buf_write=0;
    buf_read=0;

    return 1;
}

// close audio device
static void uninit(int immed)
{
    if(!immed)
	usec_sleep(get_delay() * 1000 * 1000);
    else
	waveOutReset(hWaveOut);
    while (waveOutClose(hWaveOut) == WAVERR_STILLPLAYING) usec_sleep(0);
	mp_msg(MSGT_AO, MSGL_V,"waveOut device closed\n");
    free(waveBlocks);
	mp_msg(MSGT_AO, MSGL_V,"buffer memory freed\n");
}

// stop playing and empty buffers (for seeking/pause)
static void reset(void)
{
   	waveOutReset(hWaveOut);
	buf_write=0;
	buf_read=0;
}

// stop playing, keep buffers (for pause)
static void audio_pause(void)
{
    waveOutPause(hWaveOut);
}

// resume playing, after audio_pause()
static void audio_resume(void)
{
	waveOutRestart(hWaveOut);
}

// return: how many bytes can be played without blocking
static int get_space(void)
{
    int free = buf_read - buf_write - 1;
    if (free < 0) free += BUFFER_COUNT;
    return free * BUFFER_SIZE;
}

//writes data into buffer, based on ringbuffer code in ao_sdl.c
static int write_waveOutBuffer(unsigned char* data,int len){
  WAVEHDR* current;
  int len2=0;
  int x;
  while(len>0){
    int buf_next = (buf_write + 1) % BUFFER_COUNT;
    current = &waveBlocks[buf_write];
    if(buf_next == buf_read) break;
    //unprepare the header if it is prepared
	if(current->dwFlags & WHDR_PREPARED)
           waveOutUnprepareHeader(hWaveOut, current, sizeof(WAVEHDR));
	x=BUFFER_SIZE;
    if(x>len) x=len;
    fast_memcpy(current->lpData,data+len2,x);
    len2+=x; len-=x;
	//prepare header and write data to device
	current->dwBufferLength = x;
	waveOutPrepareHeader(hWaveOut, current, sizeof(WAVEHDR));
	waveOutWrite(hWaveOut, current, sizeof(WAVEHDR));

       buf_write = buf_next;
  }
  return len2;
}

// plays 'len' bytes of 'data'
// it should round it down to outburst*n
// return: number of bytes played
static int play(void* data,int len,int flags)
{
	if (!(flags & AOPLAY_FINAL_CHUNK))
	len = (len/ao_data.outburst)*ao_data.outburst;
	return write_waveOutBuffer(data,len);
}

// return: delay in seconds between first and last sample in buffer
static float get_delay(void)
{
	int used = buf_write - buf_read;
	if (used < 0) used += BUFFER_COUNT;
	return (float)(used * BUFFER_SIZE + ao_data.buffersize)/(float)ao_data.bps;
}