annotate libao2/ao_win32.c @ 33926:6e3f257d7b05

Add disabled feature: noise filter for vo_gl.
author reimar
date Fri, 26 Aug 2011 18:51:51 +0000
parents 02b9c1a452e1
children 4e9d5dc30c00
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1 /*
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2 * Windows waveOut interface
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3 *
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4 * Copyright (c) 2002 - 2004 Sascha Sommer <saschasommer@freenet.de>
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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 * with MPlayer; if not, write to the Free Software Foundation, Inc.,
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20 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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21 */
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22
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23 #include <stdio.h>
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24 #include <stdlib.h>
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25 #include <windows.h>
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26 #include <mmsystem.h>
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27
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28 #include "config.h"
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29 #include "libaf/af_format.h"
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30 #include "audio_out.h"
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31 #include "audio_out_internal.h"
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32 #include "mp_msg.h"
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33 #include "libvo/fastmemcpy.h"
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34 #include "osdep/timer.h"
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35
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36 #define WAVE_FORMAT_DOLBY_AC3_SPDIF 0x0092
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37 #define WAVE_FORMAT_EXTENSIBLE 0xFFFE
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38
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39 static const GUID KSDATAFORMAT_SUBTYPE_PCM = {
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40 0x1,0x0000,0x0010,{0x80,0x00,0x00,0xaa,0x00,0x38,0x9b,0x71}
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41 };
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42
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43 typedef struct {
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44 WAVEFORMATEX Format;
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45 union {
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46 WORD wValidBitsPerSample;
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47 WORD wSamplesPerBlock;
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48 WORD wReserved;
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49 } Samples;
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50 DWORD dwChannelMask;
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51 GUID SubFormat;
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52 } WAVEFORMATEXTENSIBLE, *PWAVEFORMATEXTENSIBLE;
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53
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54 #define SPEAKER_FRONT_LEFT 0x1
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55 #define SPEAKER_FRONT_RIGHT 0x2
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56 #define SPEAKER_FRONT_CENTER 0x4
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57 #define SPEAKER_LOW_FREQUENCY 0x8
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58 #define SPEAKER_BACK_LEFT 0x10
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59 #define SPEAKER_BACK_RIGHT 0x20
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60 #define SPEAKER_FRONT_LEFT_OF_CENTER 0x40
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61 #define SPEAKER_FRONT_RIGHT_OF_CENTER 0x80
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62 #define SPEAKER_BACK_CENTER 0x100
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63 #define SPEAKER_SIDE_LEFT 0x200
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64 #define SPEAKER_SIDE_RIGHT 0x400
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65 #define SPEAKER_TOP_CENTER 0x800
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66 #define SPEAKER_TOP_FRONT_LEFT 0x1000
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67 #define SPEAKER_TOP_FRONT_CENTER 0x2000
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68 #define SPEAKER_TOP_FRONT_RIGHT 0x4000
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69 #define SPEAKER_TOP_BACK_LEFT 0x8000
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70 #define SPEAKER_TOP_BACK_CENTER 0x10000
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71 #define SPEAKER_TOP_BACK_RIGHT 0x20000
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72
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73 static const int channel_mask[] = {
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74 SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_LOW_FREQUENCY,
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75 SPEAKER_FRONT_LEFT | SPEAKER_FRONT_CENTER | SPEAKER_FRONT_RIGHT | SPEAKER_LOW_FREQUENCY,
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76 SPEAKER_FRONT_LEFT | SPEAKER_FRONT_CENTER | SPEAKER_FRONT_RIGHT | SPEAKER_BACK_CENTER | SPEAKER_LOW_FREQUENCY,
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77 SPEAKER_FRONT_LEFT | SPEAKER_FRONT_CENTER | SPEAKER_FRONT_RIGHT | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | SPEAKER_LOW_FREQUENCY
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78 };
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79
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80
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81
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82 #define SAMPLESIZE 1024
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83 #define BUFFER_SIZE 4096
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84 #define BUFFER_COUNT 16
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85
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86
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87 static WAVEHDR* waveBlocks; //pointer to our ringbuffer memory
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88 static HWAVEOUT hWaveOut; //handle to the waveout device
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89 static unsigned int buf_write=0;
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90 static volatile int buf_read=0;
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91
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92
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93 static const ao_info_t info =
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94 {
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95 "Windows waveOut audio output",
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96 "win32",
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97 "Sascha Sommer <saschasommer@freenet.de>",
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98 ""
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99 };
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100
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101 LIBAO_EXTERN(win32)
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102
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103 static void CALLBACK waveOutProc(HWAVEOUT hWaveOut,UINT uMsg,DWORD dwInstance,
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104 DWORD dwParam1,DWORD dwParam2)
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105 {
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106 if(uMsg != WOM_DONE)
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107 return;
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108 buf_read = (buf_read + 1) % BUFFER_COUNT;
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109 }
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110
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111 // to set/get/query special features/parameters
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112 static int control(int cmd,void *arg)
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113 {
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114 DWORD volume;
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115 switch (cmd)
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116 {
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117 case AOCONTROL_GET_VOLUME:
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118 {
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119 ao_control_vol_t* vol = (ao_control_vol_t*)arg;
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120 waveOutGetVolume(hWaveOut,&volume);
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121 vol->left = (float)(LOWORD(volume)/655.35);
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122 vol->right = (float)(HIWORD(volume)/655.35);
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123 mp_msg(MSGT_AO, MSGL_DBG2,"ao_win32: volume left:%f volume right:%f\n",vol->left,vol->right);
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124 return CONTROL_OK;
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125 }
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126 case AOCONTROL_SET_VOLUME:
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127 {
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128 ao_control_vol_t* vol = (ao_control_vol_t*)arg;
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129 volume = MAKELONG(vol->left*655.35,vol->right*655.35);
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130 waveOutSetVolume(hWaveOut,volume);
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131 return CONTROL_OK;
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132 }
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133 }
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134 return -1;
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135 }
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136
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137 // open & setup audio device
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138 // return: 1=success 0=fail
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139 static int init(int rate,int channels,int format,int flags)
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140 {
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141 WAVEFORMATEXTENSIBLE wformat;
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142 MMRESULT result;
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143 unsigned char* buffer;
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144 int i;
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145
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146 if (AF_FORMAT_IS_AC3(format))
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147 format = AF_FORMAT_AC3_NE;
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148 switch(format){
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149 case AF_FORMAT_AC3_NE:
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150 case AF_FORMAT_S24_LE:
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151 case AF_FORMAT_S16_LE:
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152 case AF_FORMAT_U8:
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153 break;
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154 default:
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155 mp_msg(MSGT_AO, MSGL_V,"ao_win32: format %s not supported defaulting to Signed 16-bit Little-Endian\n",af_fmt2str_short(format));
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156 format=AF_FORMAT_S16_LE;
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157 }
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158
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159 // FIXME multichannel mode is buggy
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160 if(channels > 2)
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161 channels = 2;
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162
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163 //fill global ao_data
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164 ao_data.channels=channels;
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165 ao_data.samplerate=rate;
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166 ao_data.format=format;
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167 ao_data.bps=channels*rate;
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168 if(format != AF_FORMAT_U8 && format != AF_FORMAT_S8)
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169 ao_data.bps*=2;
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170 ao_data.outburst = BUFFER_SIZE;
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171 if(ao_data.buffersize==-1)
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172 {
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173 ao_data.buffersize=af_fmt2bits(format)/8;
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174 ao_data.buffersize*= channels;
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175 ao_data.buffersize*= SAMPLESIZE;
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176 }
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177 mp_msg(MSGT_AO, MSGL_V,"ao_win32: Samplerate:%iHz Channels:%i Format:%s\n",rate, channels, af_fmt2str_short(format));
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178 mp_msg(MSGT_AO, MSGL_V,"ao_win32: Buffersize:%d\n",ao_data.buffersize);
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179
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180 //fill waveformatex
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181 ZeroMemory( &wformat, sizeof(WAVEFORMATEXTENSIBLE));
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182 wformat.Format.cbSize = (channels>2)?sizeof(WAVEFORMATEXTENSIBLE)-sizeof(WAVEFORMATEX):0;
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183 wformat.Format.nChannels = channels;
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184 wformat.Format.nSamplesPerSec = rate;
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185 if(AF_FORMAT_IS_AC3(format))
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186 {
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187 wformat.Format.wFormatTag = WAVE_FORMAT_DOLBY_AC3_SPDIF;
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188 wformat.Format.wBitsPerSample = 16;
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189 wformat.Format.nBlockAlign = 4;
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190 }
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191 else
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192 {
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193 wformat.Format.wFormatTag = (channels>2)?WAVE_FORMAT_EXTENSIBLE:WAVE_FORMAT_PCM;
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194 wformat.Format.wBitsPerSample = af_fmt2bits(format);
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195 wformat.Format.nBlockAlign = wformat.Format.nChannels * (wformat.Format.wBitsPerSample >> 3);
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196 }
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197 if(channels>2)
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198 {
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199 wformat.dwChannelMask = channel_mask[channels-3];
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200 wformat.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
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201 wformat.Samples.wValidBitsPerSample=af_fmt2bits(format);
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202 }
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203
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204 wformat.Format.nAvgBytesPerSec = wformat.Format.nSamplesPerSec * wformat.Format.nBlockAlign;
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205
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206 //open sound device
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207 //WAVE_MAPPER always points to the default wave device on the system
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208 result = waveOutOpen(&hWaveOut,WAVE_MAPPER,(WAVEFORMATEX*)&wformat,(DWORD_PTR)waveOutProc,0,CALLBACK_FUNCTION);
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209 if(result == WAVERR_BADFORMAT)
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210 {
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211 mp_msg(MSGT_AO, MSGL_ERR,"ao_win32: format not supported switching to default\n");
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212 ao_data.channels = wformat.Format.nChannels = 2;
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213 ao_data.samplerate = wformat.Format.nSamplesPerSec = 44100;
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214 ao_data.format = AF_FORMAT_S16_LE;
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215 ao_data.bps=ao_data.channels * ao_data.samplerate*2;
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216 wformat.Format.wBitsPerSample=16;
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217 wformat.Format.wFormatTag=WAVE_FORMAT_PCM;
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218 wformat.Format.nBlockAlign = wformat.Format.nChannels * (wformat.Format.wBitsPerSample >> 3);
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219 wformat.Format.nAvgBytesPerSec = wformat.Format.nSamplesPerSec * wformat.Format.nBlockAlign;
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220 ao_data.buffersize=(wformat.Format.wBitsPerSample>>3)*wformat.Format.nChannels*SAMPLESIZE;
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221 result = waveOutOpen(&hWaveOut,WAVE_MAPPER,(WAVEFORMATEX*)&wformat,(DWORD_PTR)waveOutProc,0,CALLBACK_FUNCTION);
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222 }
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223 if(result != MMSYSERR_NOERROR)
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224 {
14503
af82f404a979 print why waveOutOpen failed
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225 mp_msg(MSGT_AO, MSGL_ERR,"ao_win32: unable to open wave mapper device (result=%i)\n",result);
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226 return 0;
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227 }
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228 //allocate buffer memory as one big block
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229 buffer = calloc(BUFFER_COUNT, BUFFER_SIZE + sizeof(WAVEHDR));
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230 //and setup pointers to each buffer
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231 waveBlocks = (WAVEHDR*)buffer;
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232 buffer += sizeof(WAVEHDR) * BUFFER_COUNT;
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233 for(i = 0; i < BUFFER_COUNT; i++) {
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234 waveBlocks[i].lpData = buffer;
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235 buffer += BUFFER_SIZE;
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236 }
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237 buf_write=0;
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238 buf_read=0;
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239
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240 return 1;
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241 }
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242
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243 // close audio device
12145
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parents: 12093
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244 static void uninit(int immed)
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245 {
28793
afbd8dd23299 Sleep based on get_delay in ao_win32 uninit instead of a loop.
reimar
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246 if(!immed)
afbd8dd23299 Sleep based on get_delay in ao_win32 uninit instead of a loop.
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247 usec_sleep(get_delay() * 1000 * 1000);
28883
c3f295ceae51 Do not call waveOutReset in uninit if you should wait till playing finishes,
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248 else
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249 waveOutReset(hWaveOut);
28883
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250 while (waveOutClose(hWaveOut) == WAVERR_STILLPLAYING) usec_sleep(0);
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251 mp_msg(MSGT_AO, MSGL_V,"waveOut device closed\n");
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252 free(waveBlocks);
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253 mp_msg(MSGT_AO, MSGL_V,"buffer memory freed\n");
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254 }
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255
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256 // stop playing and empty buffers (for seeking/pause)
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257 static void reset(void)
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258 {
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259 waveOutReset(hWaveOut);
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260 buf_write=0;
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261 buf_read=0;
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262 }
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263
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264 // stop playing, keep buffers (for pause)
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265 static void audio_pause(void)
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266 {
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267 waveOutPause(hWaveOut);
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268 }
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269
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270 // resume playing, after audio_pause()
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271 static void audio_resume(void)
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272 {
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273 waveOutRestart(hWaveOut);
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274 }
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275
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276 // return: how many bytes can be played without blocking
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277 static int get_space(void)
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278 {
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279 int free = buf_read - buf_write - 1;
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280 if (free < 0) free += BUFFER_COUNT;
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281 return free * BUFFER_SIZE;
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282 }
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283
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284 //writes data into buffer, based on ringbuffer code in ao_sdl.c
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285 static int write_waveOutBuffer(unsigned char* data,int len){
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286 WAVEHDR* current;
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287 int len2=0;
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288 int x;
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289 while(len>0){
28822
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290 int buf_next = (buf_write + 1) % BUFFER_COUNT;
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291 current = &waveBlocks[buf_write];
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292 if(buf_next == buf_read) break;
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293 //unprepare the header if it is prepared
28821
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294 if(current->dwFlags & WHDR_PREPARED)
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295 waveOutUnprepareHeader(hWaveOut, current, sizeof(WAVEHDR));
28821
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296 x=BUFFER_SIZE;
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297 if(x>len) x=len;
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298 fast_memcpy(current->lpData,data+len2,x);
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299 len2+=x; len-=x;
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300 //prepare header and write data to device
28818
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parents: 28817
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301 current->dwBufferLength = x;
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302 waveOutPrepareHeader(hWaveOut, current, sizeof(WAVEHDR));
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303 waveOutWrite(hWaveOut, current, sizeof(WAVEHDR));
28821
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diff changeset
304
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diff changeset
305 buf_write = buf_next;
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306 }
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307 return len2;
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308 }
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309
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310 // plays 'len' bytes of 'data'
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311 // it should round it down to outburst*n
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312 // return: number of bytes played
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313 static int play(void* data,int len,int flags)
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314 {
20250
7cfd3a04d537 Implement AOPLAY_FINAL_CHUNK support for dshow and win32 out.
reimar
parents: 19835
diff changeset
315 if (!(flags & AOPLAY_FINAL_CHUNK))
12151
75fdb659f5bf round len to outburst and increment full_buffers at the correct time, patch by Nehal <nehalmistry at gmx.net>
faust3
parents: 12146
diff changeset
316 len = (len/ao_data.outburst)*ao_data.outburst;
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317 return write_waveOutBuffer(data,len);
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318 }
10106
b5b7b2f4f069 10l remove unused global
faust3
parents: 9633
diff changeset
319
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320 // return: delay in seconds between first and last sample in buffer
18915
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321 static float get_delay(void)
7913
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322 {
28822
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323 int used = buf_write - buf_read;
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324 if (used < 0) used += BUFFER_COUNT;
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325 return (float)(used * BUFFER_SIZE + ao_data.buffersize)/(float)ao_data.bps;
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326 }