annotate libao2/ao_win32.c @ 13963:a22ffcf85c53

fixed missing para tag
author gabrov
date Wed, 17 Nov 2004 16:17:47 +0000
parents 83c5f9888576
children a92101a7eb49
Ignore whitespace changes - Everywhere: Within whitespace: At end of lines:
rev   line source
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1 /******************************************************************************
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2 * ao_win32.c: Windows waveOut interface for MPlayer
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3 * Copyright (c) 2002 - 2004 Sascha Sommer <saschasommer@freenet.de>.
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4 *
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5 * This program is free software; you can redistribute it and/or modify
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6 * it under the terms of the GNU General Public License as published by
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7 * the Free Software Foundation; either version 2 of the License, or
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8 * (at your option) any later version.
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9 *
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10 * This program is distributed in the hope that it will be useful,
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11 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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13 * GNU General Public License for more details.
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14 *
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15 * You should have received a copy of the GNU General Public License
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16 * along with this program; if not, write to the Free Software
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17 * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111, USA.
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18 *
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19 *****************************************************************************/
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20
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21 #include <stdio.h>
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22 #include <stdlib.h>
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23 #include <windows.h>
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24 #include <mmsystem.h>
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25
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26 #include "afmt.h"
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27 #include "audio_out.h"
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28 #include "audio_out_internal.h"
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29 #include "../mp_msg.h"
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30 #include "../libvo/fastmemcpy.h"
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31 #include "osdep/timer.h"
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32
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33 #define WAVE_FORMAT_DOLBY_AC3_SPDIF 0x0092
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34 #define WAVE_FORMAT_EXTENSIBLE 0xFFFE
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35
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36 static const GUID KSDATAFORMAT_SUBTYPE_PCM = {
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37 0x1,0x0000,0x0010,{0x80,0x00,0x00,0xaa,0x00,0x38,0x9b,0x71}
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38 };
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39
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40 typedef struct {
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41 WAVEFORMATEX Format;
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42 union {
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43 WORD wValidBitsPerSample;
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44 WORD wSamplesPerBlock;
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45 WORD wReserved;
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46 } Samples;
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47 DWORD dwChannelMask;
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48 GUID SubFormat;
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49 } WAVEFORMATEXTENSIBLE, *PWAVEFORMATEXTENSIBLE;
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50
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51 #define SPEAKER_FRONT_LEFT 0x1
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52 #define SPEAKER_FRONT_RIGHT 0x2
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53 #define SPEAKER_FRONT_CENTER 0x4
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54 #define SPEAKER_LOW_FREQUENCY 0x8
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55 #define SPEAKER_BACK_LEFT 0x10
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56 #define SPEAKER_BACK_RIGHT 0x20
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57 #define SPEAKER_FRONT_LEFT_OF_CENTER 0x40
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58 #define SPEAKER_FRONT_RIGHT_OF_CENTER 0x80
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59 #define SPEAKER_BACK_CENTER 0x100
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60 #define SPEAKER_SIDE_LEFT 0x200
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61 #define SPEAKER_SIDE_RIGHT 0x400
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62 #define SPEAKER_TOP_CENTER 0x800
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63 #define SPEAKER_TOP_FRONT_LEFT 0x1000
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64 #define SPEAKER_TOP_FRONT_CENTER 0x2000
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65 #define SPEAKER_TOP_FRONT_RIGHT 0x4000
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66 #define SPEAKER_TOP_BACK_LEFT 0x8000
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67 #define SPEAKER_TOP_BACK_CENTER 0x10000
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68 #define SPEAKER_TOP_BACK_RIGHT 0x20000
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69
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70 static const int channel_mask[] = {
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71 SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_LOW_FREQUENCY,
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72 SPEAKER_FRONT_LEFT | SPEAKER_FRONT_CENTER | SPEAKER_FRONT_RIGHT | SPEAKER_LOW_FREQUENCY,
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73 SPEAKER_FRONT_LEFT | SPEAKER_FRONT_CENTER | SPEAKER_FRONT_RIGHT | SPEAKER_BACK_CENTER | SPEAKER_LOW_FREQUENCY,
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74 SPEAKER_FRONT_LEFT | SPEAKER_FRONT_CENTER | SPEAKER_FRONT_RIGHT | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | SPEAKER_LOW_FREQUENCY
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75 };
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76
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77
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78
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79 #define SAMPLESIZE 1024
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80 #define BUFFER_SIZE 4096
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81 #define BUFFER_COUNT 16
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82
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83
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84 static WAVEHDR* waveBlocks; //pointer to our ringbuffer memory
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85 static HWAVEOUT hWaveOut; //handle to the waveout device
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86 static unsigned int buf_write=0;
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87 static unsigned int buf_write_pos=0;
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88 static int full_buffers=0;
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89 static int buffered_bytes=0;
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90
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91
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92 static ao_info_t info =
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93 {
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94 "Windows waveOut audio output",
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95 "win32",
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96 "Sascha Sommer <saschasommer@freenet.de>",
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97 ""
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98 };
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99
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100 LIBAO_EXTERN(win32)
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101
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102 static void CALLBACK waveOutProc(HWAVEOUT hWaveOut,UINT uMsg,DWORD dwInstance,
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103 DWORD dwParam1,DWORD dwParam2)
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104 {
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105 if(uMsg != WOM_DONE)
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106 return;
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107 if (full_buffers) {
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108 buffered_bytes-=BUFFER_SIZE;
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109 --full_buffers;
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110 } else {
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111 buffered_bytes=0;
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112 }
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113 }
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114
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115 // to set/get/query special features/parameters
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116 static int control(int cmd,void *arg)
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117 {
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118 DWORD volume;
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119 switch (cmd)
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120 {
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121 case AOCONTROL_GET_VOLUME:
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122 {
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123 ao_control_vol_t* vol = (ao_control_vol_t*)arg;
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124 waveOutGetVolume(hWaveOut,&volume);
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125 vol->left = (float)(LOWORD(volume)/655.35);
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126 vol->right = (float)(HIWORD(volume)/655.35);
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127 mp_msg(MSGT_AO, MSGL_DBG2,"ao_win32: volume left:%f volume right:%f\n",vol->left,vol->right);
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128 return CONTROL_OK;
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129 }
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130 case AOCONTROL_SET_VOLUME:
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131 {
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132 ao_control_vol_t* vol = (ao_control_vol_t*)arg;
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133 volume = MAKELONG(vol->left*655.35,vol->right*655.35);
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134 waveOutSetVolume(hWaveOut,volume);
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135 return CONTROL_OK;
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136 }
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137 }
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138 return -1;
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139 }
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140
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141 // open & setup audio device
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142 // return: 1=success 0=fail
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143 static int init(int rate,int channels,int format,int flags)
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144 {
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145 WAVEFORMATEXTENSIBLE wformat;
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146 DWORD totalBufferSize = (BUFFER_SIZE + sizeof(WAVEHDR)) * BUFFER_COUNT;
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147 MMRESULT result;
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148 unsigned char* buffer;
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149 int i;
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150
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151 //fill global ao_data
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152 ao_data.channels=channels;
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153 ao_data.samplerate=rate;
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154 ao_data.format=format;
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155 ao_data.bps=channels*rate;
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156 if(format != AFMT_U8 && format != AFMT_S8)
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157 ao_data.bps*=2;
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158 if(ao_data.buffersize==-1)
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159 {
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160 ao_data.buffersize=audio_out_format_bits(format)/8;
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161 ao_data.buffersize*= channels;
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162 ao_data.buffersize*= SAMPLESIZE;
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163 }
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164 mp_msg(MSGT_AO, MSGL_V,"ao_win32: Samplerate:%iHz Channels:%i Format:%s\n",rate, channels, audio_out_format_name(format));
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165 mp_msg(MSGT_AO, MSGL_V,"ao_win32: Buffersize:%d\n",ao_data.buffersize);
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166
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167 //fill waveformatex
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168 ZeroMemory( &wformat, sizeof(WAVEFORMATEXTENSIBLE));
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169 wformat.Format.cbSize = (channels>2)?sizeof(WAVEFORMATEXTENSIBLE)-sizeof(WAVEFORMATEX):0;
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170 wformat.Format.nChannels = channels;
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171 wformat.Format.nSamplesPerSec = rate;
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172 if(format == AFMT_AC3)
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173 {
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174 wformat.Format.wFormatTag = WAVE_FORMAT_DOLBY_AC3_SPDIF;
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175 wformat.Format.wBitsPerSample = 16;
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176 wformat.Format.nBlockAlign = 4;
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177 }
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178 else
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179 {
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180 wformat.Format.wFormatTag = (channels>2)?WAVE_FORMAT_EXTENSIBLE:WAVE_FORMAT_PCM;
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181 wformat.Format.wBitsPerSample = audio_out_format_bits(format);
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182 wformat.Format.nBlockAlign = wformat.Format.nChannels * (wformat.Format.wBitsPerSample >> 3);
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183 }
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184 if(channels>2)
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185 {
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186 wformat.dwChannelMask = channel_mask[channels-3];
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187 wformat.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
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188 wformat.Samples.wValidBitsPerSample=audio_out_format_bits(format);
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189 }
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190
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191 wformat.Format.nAvgBytesPerSec = wformat.Format.nSamplesPerSec * wformat.Format.nBlockAlign;
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192
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193 //open sound device
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194 //WAVE_MAPPER always points to the default wave device on the system
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195 result = waveOutOpen(&hWaveOut,WAVE_MAPPER,(WAVEFORMATEX*)&wformat,(DWORD_PTR)waveOutProc,0,CALLBACK_FUNCTION|WAVE_FORMAT_DIRECT);
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196 if(result == WAVERR_BADFORMAT)
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197 {
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198 mp_msg(MSGT_AO, MSGL_ERR,"ao_win32: format not supported switching to default\n");
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199 ao_data.channels = wformat.Format.nChannels = 2;
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200 ao_data.samplerate = wformat.Format.nSamplesPerSec = 44100;
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201 ao_data.format = AFMT_S16_LE;
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202 ao_data.bps=ao_data.channels * ao_data.samplerate*2;
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203 wformat.Format.wBitsPerSample=16;
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204 wformat.Format.wFormatTag=WAVE_FORMAT_PCM;
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205 wformat.Format.nBlockAlign = wformat.Format.nChannels * (wformat.Format.wBitsPerSample >> 3);
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206 wformat.Format.nAvgBytesPerSec = wformat.Format.nSamplesPerSec * wformat.Format.nBlockAlign;
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207 ao_data.buffersize=(wformat.Format.wBitsPerSample>>3)*wformat.Format.nChannels*SAMPLESIZE;
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208 result = waveOutOpen(&hWaveOut,WAVE_MAPPER,(WAVEFORMATEX*)&wformat,(DWORD_PTR)waveOutProc,0,CALLBACK_FUNCTION);
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209 }
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210 if(result != MMSYSERR_NOERROR)
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211 {
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212 mp_msg(MSGT_AO, MSGL_ERR,"ao_win32: unable to open wave mapper device\n");
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213 return 0;
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214 }
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215 //allocate buffer memory as one big block
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216 buffer = malloc(totalBufferSize);
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217 memset(buffer,0x0,totalBufferSize);
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218 //and setup pointers to each buffer
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219 waveBlocks = (WAVEHDR*)buffer;
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220 buffer += sizeof(WAVEHDR) * BUFFER_COUNT;
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221 for(i = 0; i < BUFFER_COUNT; i++) {
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222 waveBlocks[i].lpData = buffer;
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223 buffer += BUFFER_SIZE;
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224 }
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225 buf_write=0;
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226 buf_write_pos=0;
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227 full_buffers=0;
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228 buffered_bytes=0;
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229
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230 return 1;
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231 }
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232
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233 // close audio device
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234 static void uninit(int immed)
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235 {
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236 if(!immed)while(buffered_bytes > 0)usec_sleep(50000);
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237 else buffered_bytes=0;
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238 waveOutReset(hWaveOut);
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239 waveOutClose(hWaveOut);
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240 mp_msg(MSGT_AO, MSGL_V,"waveOut device closed\n");
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241 free(waveBlocks);
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242 mp_msg(MSGT_AO, MSGL_V,"buffer memory freed\n");
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243 }
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244
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245 // stop playing and empty buffers (for seeking/pause)
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246 static void reset()
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247 {
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248 waveOutReset(hWaveOut);
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249 buf_write=0;
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250 buf_write_pos=0;
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251 full_buffers=0;
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252 buffered_bytes=0;
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253 }
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254
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255 // stop playing, keep buffers (for pause)
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256 static void audio_pause()
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257 {
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258 waveOutPause(hWaveOut);
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259 }
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260
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261 // resume playing, after audio_pause()
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262 static void audio_resume()
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263 {
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264 waveOutRestart(hWaveOut);
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265 }
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266
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267 // return: how many bytes can be played without blocking
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268 static int get_space()
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269 {
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270 return BUFFER_COUNT*BUFFER_SIZE - buffered_bytes;
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271 }
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272
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273 //writes data into buffer, based on ringbuffer code in ao_sdl.c
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274 static int write_waveOutBuffer(unsigned char* data,int len){
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275 WAVEHDR* current;
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276 int len2=0;
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277 int x;
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278 while(len>0){
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279 current = &waveBlocks[buf_write];
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280 if(buffered_bytes==BUFFER_COUNT*BUFFER_SIZE) break;
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281 //unprepare the header if it is prepared
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282 if(current->dwFlags & WHDR_PREPARED)
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283 waveOutUnprepareHeader(hWaveOut, current, sizeof(WAVEHDR));
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284 x=BUFFER_SIZE-buf_write_pos;
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285 if(x>len) x=len;
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286 memcpy(current->lpData+buf_write_pos,data+len2,x);
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287 if(buf_write_pos==0)full_buffers++;
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288 len2+=x; len-=x;
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289 buffered_bytes+=x; buf_write_pos+=x;
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290 //prepare header and write data to device
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291 current->dwBufferLength = buf_write_pos;
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292 waveOutPrepareHeader(hWaveOut, current, sizeof(WAVEHDR));
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293 waveOutWrite(hWaveOut, current, sizeof(WAVEHDR));
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294
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295 if(buf_write_pos>=BUFFER_SIZE){ //buffer is full find next
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296 // block is full, find next!
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297 buf_write=(buf_write+1)%BUFFER_COUNT;
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298 buf_write_pos=0;
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299 }
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300 }
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301 return len2;
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302 }
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303
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304 // plays 'len' bytes of 'data'
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305 // it should round it down to outburst*n
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306 // return: number of bytes played
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307 static int play(void* data,int len,int flags)
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308 {
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309 len = (len/ao_data.outburst)*ao_data.outburst;
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310 return write_waveOutBuffer(data,len);
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311 }
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312
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313 // return: delay in seconds between first and last sample in buffer
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314 static float get_delay()
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315 {
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316 return (float)(buffered_bytes + ao_data.buffersize)/(float)ao_data.bps;
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317 }