Mercurial > mplayer.hg
annotate libao2/ao_dsound.c @ 21715:9d96f4df253a
sync with r21655, based on a file conversion of help_mp-zh_CN.h
file provided by Eric Wang, eric7wang gmail com
author | kraymer |
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date | Fri, 22 Dec 2006 22:56:07 +0000 |
parents | 7cfd3a04d537 |
children | a124f3abc1ec |
rev | line source |
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1 /****************************************************************************** |
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2 * ao_dsound.c: Windows DirectSound interface for MPlayer |
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3 * Copyright (c) 2004 Gabor Szecsi <deje@miki.hu> |
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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 |
19614 | 17 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
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18 * |
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19 *****************************************************************************/ |
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20 /** |
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21 \todo verify/extend multichannel support |
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22 */ |
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23 |
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24 |
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25 #include <stdio.h> |
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26 #include <stdlib.h> |
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27 #include <windows.h> |
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28 #define DIRECTSOUND_VERSION 0x0600 |
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29 #include <dsound.h> |
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30 #include <math.h> |
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31 |
14479 | 32 #include "config.h" |
14245 | 33 #include "libaf/af_format.h" |
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34 #include "audio_out.h" |
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35 #include "audio_out_internal.h" |
14123 | 36 #include "mp_msg.h" |
37 #include "libvo/fastmemcpy.h" | |
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38 #include "osdep/timer.h" |
14398 | 39 #include "subopt-helper.h" |
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40 |
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41 |
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42 static ao_info_t info = |
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43 { |
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44 "Windows DirectSound audio output", |
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45 "dsound", |
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46 "Gabor Szecsi <deje@miki.hu>", |
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47 "" |
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48 }; |
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49 |
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50 LIBAO_EXTERN(dsound) |
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51 |
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52 /** |
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53 \todo use the definitions from the win32 api headers when they define these |
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54 */ |
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55 #if 1 |
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56 #define WAVE_FORMAT_IEEE_FLOAT 0x0003 |
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57 #define WAVE_FORMAT_DOLBY_AC3_SPDIF 0x0092 |
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58 #define WAVE_FORMAT_EXTENSIBLE 0xFFFE |
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59 |
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60 static const GUID KSDATAFORMAT_SUBTYPE_PCM = {0x1,0x0000,0x0010, {0x80,0x00,0x00,0xaa,0x00,0x38,0x9b,0x71}}; |
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61 |
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62 #define SPEAKER_FRONT_LEFT 0x1 |
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63 #define SPEAKER_FRONT_RIGHT 0x2 |
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64 #define SPEAKER_FRONT_CENTER 0x4 |
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65 #define SPEAKER_LOW_FREQUENCY 0x8 |
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66 #define SPEAKER_BACK_LEFT 0x10 |
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67 #define SPEAKER_BACK_RIGHT 0x20 |
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68 #define SPEAKER_FRONT_LEFT_OF_CENTER 0x40 |
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69 #define SPEAKER_FRONT_RIGHT_OF_CENTER 0x80 |
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70 #define SPEAKER_BACK_CENTER 0x100 |
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71 #define SPEAKER_SIDE_LEFT 0x200 |
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72 #define SPEAKER_SIDE_RIGHT 0x400 |
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73 #define SPEAKER_TOP_CENTER 0x800 |
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74 #define SPEAKER_TOP_FRONT_LEFT 0x1000 |
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75 #define SPEAKER_TOP_FRONT_CENTER 0x2000 |
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76 #define SPEAKER_TOP_FRONT_RIGHT 0x4000 |
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77 #define SPEAKER_TOP_BACK_LEFT 0x8000 |
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78 #define SPEAKER_TOP_BACK_CENTER 0x10000 |
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79 #define SPEAKER_TOP_BACK_RIGHT 0x20000 |
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80 #define SPEAKER_RESERVED 0x80000000 |
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81 |
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82 #define DSSPEAKER_HEADPHONE 0x00000001 |
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83 #define DSSPEAKER_MONO 0x00000002 |
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84 #define DSSPEAKER_QUAD 0x00000003 |
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85 #define DSSPEAKER_STEREO 0x00000004 |
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86 #define DSSPEAKER_SURROUND 0x00000005 |
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87 #define DSSPEAKER_5POINT1 0x00000006 |
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88 |
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89 #ifndef _WAVEFORMATEXTENSIBLE_ |
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90 typedef struct { |
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91 WAVEFORMATEX Format; |
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92 union { |
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93 WORD wValidBitsPerSample; /* bits of precision */ |
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94 WORD wSamplesPerBlock; /* valid if wBitsPerSample==0 */ |
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95 WORD wReserved; /* If neither applies, set to zero. */ |
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96 } Samples; |
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97 DWORD dwChannelMask; /* which channels are */ |
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98 /* present in stream */ |
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99 GUID SubFormat; |
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100 } WAVEFORMATEXTENSIBLE, *PWAVEFORMATEXTENSIBLE; |
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101 #endif |
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102 |
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103 #endif |
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104 |
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105 static const int channel_mask[] = { |
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106 SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_LOW_FREQUENCY, |
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107 SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT, |
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108 SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | SPEAKER_LOW_FREQUENCY, |
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109 SPEAKER_FRONT_LEFT | SPEAKER_FRONT_CENTER | SPEAKER_FRONT_RIGHT | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | SPEAKER_LOW_FREQUENCY |
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110 }; |
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111 |
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112 static HINSTANCE hdsound_dll = NULL; ///handle to the dll |
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113 static LPDIRECTSOUND hds = NULL; ///direct sound object |
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114 static LPDIRECTSOUNDBUFFER hdspribuf = NULL; ///primary direct sound buffer |
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115 static LPDIRECTSOUNDBUFFER hdsbuf = NULL; ///secondary direct sound buffer (stream buffer) |
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116 static int buffer_size = 0; ///size in bytes of the direct sound buffer |
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117 static int write_offset = 0; ///offset of the write cursor in the direct sound buffer |
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118 static int min_free_space = 0; ///if the free space is below this value get_space() will return 0 |
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119 ///there will always be at least this amout of free space to prevent |
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120 ///get_space() from returning wrong values when buffer is 100% full. |
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121 ///will be replaced with nBlockAlign in init() |
14398 | 122 static int device_num = 0; ///wanted device number |
123 static GUID device; ///guid of the device | |
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124 |
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125 /***************************************************************************************/ |
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126 |
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127 /** |
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128 \brief output error message |
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129 \param err error code |
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130 \return string with the error message |
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131 */ |
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132 static char * dserr2str(int err) |
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133 { |
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134 switch (err) { |
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135 case DS_OK: return "DS_OK"; |
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136 case DS_NO_VIRTUALIZATION: return "DS_NO_VIRTUALIZATION"; |
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137 case DSERR_ALLOCATED: return "DS_NO_VIRTUALIZATION"; |
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138 case DSERR_CONTROLUNAVAIL: return "DSERR_CONTROLUNAVAIL"; |
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139 case DSERR_INVALIDPARAM: return "DSERR_INVALIDPARAM"; |
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140 case DSERR_INVALIDCALL: return "DSERR_INVALIDCALL"; |
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141 case DSERR_GENERIC: return "DSERR_GENERIC"; |
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142 case DSERR_PRIOLEVELNEEDED: return "DSERR_PRIOLEVELNEEDED"; |
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143 case DSERR_OUTOFMEMORY: return "DSERR_OUTOFMEMORY"; |
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144 case DSERR_BADFORMAT: return "DSERR_BADFORMAT"; |
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145 case DSERR_UNSUPPORTED: return "DSERR_UNSUPPORTED"; |
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146 case DSERR_NODRIVER: return "DSERR_NODRIVER"; |
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147 case DSERR_ALREADYINITIALIZED: return "DSERR_ALREADYINITIALIZED"; |
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148 case DSERR_NOAGGREGATION: return "DSERR_NOAGGREGATION"; |
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149 case DSERR_BUFFERLOST: return "DSERR_BUFFERLOST"; |
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150 case DSERR_OTHERAPPHASPRIO: return "DSERR_OTHERAPPHASPRIO"; |
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151 case DSERR_UNINITIALIZED: return "DSERR_UNINITIALIZED"; |
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152 case DSERR_NOINTERFACE: return "DSERR_NOINTERFACE"; |
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153 case DSERR_ACCESSDENIED: return "DSERR_ACCESSDENIED"; |
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154 default: return "unknown"; |
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155 } |
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156 } |
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157 |
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158 /** |
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159 \brief uninitialize direct sound |
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160 */ |
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161 static void UninitDirectSound(void) |
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162 { |
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163 // finally release the DirectSound object |
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164 if (hds) { |
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165 IDirectSound_Release(hds); |
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166 hds = NULL; |
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167 } |
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168 // free DSOUND.DLL |
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169 if (hdsound_dll) { |
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170 FreeLibrary(hdsound_dll); |
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171 hdsound_dll = NULL; |
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172 } |
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173 mp_msg(MSGT_AO, MSGL_V, "ao_dsound: DirectSound uninitialized\n"); |
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174 } |
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175 |
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176 /** |
14398 | 177 \brief print the commandline help |
178 */ | |
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179 static void print_help(void) |
14398 | 180 { |
181 mp_msg(MSGT_AO, MSGL_FATAL, | |
182 "\n-ao dsound commandline help:\n" | |
183 "Example: mplayer -ao dsound:device=1\n" | |
184 " sets 1st device\n" | |
185 "\nOptions:\n" | |
186 " device=<device-number>\n" | |
187 " Sets device number, use -v to get a list\n"); | |
188 } | |
189 | |
190 | |
191 /** | |
192 \brief enumerate direct sound devices | |
193 \return TRUE to continue with the enumeration | |
194 */ | |
195 static BOOL CALLBACK DirectSoundEnum(LPGUID guid,LPCSTR desc,LPCSTR module,LPVOID context) | |
196 { | |
197 int* device_index=context; | |
198 mp_msg(MSGT_AO, MSGL_V,"%i %s ",*device_index,desc); | |
199 if(device_num==*device_index){ | |
200 mp_msg(MSGT_AO, MSGL_V,"<--"); | |
201 if(guid){ | |
202 memcpy(&device,guid,sizeof(GUID)); | |
203 } | |
204 } | |
205 mp_msg(MSGT_AO, MSGL_V,"\n"); | |
206 (*device_index)++; | |
207 return TRUE; | |
208 } | |
209 | |
210 | |
211 /** | |
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212 \brief initilize direct sound |
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213 \return 0 if error, 1 if ok |
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214 */ |
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215 static int InitDirectSound(void) |
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216 { |
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217 DSCAPS dscaps; |
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218 |
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219 // initialize directsound |
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220 HRESULT (WINAPI *OurDirectSoundCreate)(LPGUID, LPDIRECTSOUND *, LPUNKNOWN); |
14398 | 221 HRESULT (WINAPI *OurDirectSoundEnumerate)(LPDSENUMCALLBACKA, LPVOID); |
222 int device_index=0; | |
223 opt_t subopts[] = { | |
224 {"device", OPT_ARG_INT, &device_num,NULL}, | |
225 {NULL} | |
226 }; | |
227 if (subopt_parse(ao_subdevice, subopts) != 0) { | |
228 print_help(); | |
229 return 0; | |
230 } | |
231 | |
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232 hdsound_dll = LoadLibrary("DSOUND.DLL"); |
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233 if (hdsound_dll == NULL) { |
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234 mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: cannot load DSOUND.DLL\n"); |
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235 return 0; |
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236 } |
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237 OurDirectSoundCreate = (void*)GetProcAddress(hdsound_dll, "DirectSoundCreate"); |
14398 | 238 OurDirectSoundEnumerate = (void*)GetProcAddress(hdsound_dll, "DirectSoundEnumerateA"); |
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239 |
14398 | 240 if (OurDirectSoundCreate == NULL || OurDirectSoundEnumerate == NULL) { |
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241 mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: GetProcAddress FAILED\n"); |
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242 FreeLibrary(hdsound_dll); |
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243 return 0; |
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244 } |
14398 | 245 |
246 // Enumerate all directsound devices | |
247 mp_msg(MSGT_AO, MSGL_V,"ao_dsound: Output Devices:\n"); | |
248 OurDirectSoundEnumerate(DirectSoundEnum,&device_index); | |
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249 |
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250 // Create the direct sound object |
14398 | 251 if FAILED(OurDirectSoundCreate((device_num)?&device:NULL, &hds, NULL )) { |
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252 mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: cannot create a DirectSound device\n"); |
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253 FreeLibrary(hdsound_dll); |
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254 return 0; |
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255 } |
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256 |
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257 /* Set DirectSound Cooperative level, ie what control we want over Windows |
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258 * sound device. In our case, DSSCL_EXCLUSIVE means that we can modify the |
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259 * settings of the primary buffer, but also that only the sound of our |
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260 * application will be hearable when it will have the focus. |
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261 * !!! (this is not really working as intended yet because to set the |
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262 * cooperative level you need the window handle of your application, and |
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263 * I don't know of any easy way to get it. Especially since we might play |
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264 * sound without any video, and so what window handle should we use ??? |
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265 * The hack for now is to use the Desktop window handle - it seems to be |
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266 * working */ |
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267 if (IDirectSound_SetCooperativeLevel(hds, GetDesktopWindow(), DSSCL_EXCLUSIVE)) { |
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268 mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: cannot set direct sound cooperative level\n"); |
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269 IDirectSound_Release(hds); |
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270 FreeLibrary(hdsound_dll); |
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271 return 0; |
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272 } |
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273 mp_msg(MSGT_AO, MSGL_V, "ao_dsound: DirectSound initialized\n"); |
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274 |
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275 memset(&dscaps, 0, sizeof(DSCAPS)); |
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276 dscaps.dwSize = sizeof(DSCAPS); |
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277 if (DS_OK == IDirectSound_GetCaps(hds, &dscaps)) { |
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278 if (dscaps.dwFlags & DSCAPS_EMULDRIVER) mp_msg(MSGT_AO, MSGL_V, "ao_dsound: DirectSound is emulated, waveOut may give better performance\n"); |
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279 } else { |
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280 mp_msg(MSGT_AO, MSGL_V, "ao_dsound: cannot get device capabilities\n"); |
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281 } |
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282 |
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283 return 1; |
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284 } |
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285 |
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286 /** |
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287 \brief destroy the direct sound buffer |
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288 */ |
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289 static void DestroyBuffer(void) |
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290 { |
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291 if (hdsbuf) { |
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292 IDirectSoundBuffer_Release(hdsbuf); |
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293 hdsbuf = NULL; |
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294 } |
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295 if (hdspribuf) { |
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296 IDirectSoundBuffer_Release(hdspribuf); |
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297 hdspribuf = NULL; |
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298 } |
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299 } |
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300 |
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301 /** |
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302 \brief fill sound buffer |
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303 \param data pointer to the sound data to copy |
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304 \param len length of the data to copy in bytes |
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305 \return number of copyed bytes |
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306 */ |
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307 static int write_buffer(unsigned char *data, int len) |
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308 { |
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309 HRESULT res; |
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310 LPVOID lpvPtr1; |
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311 DWORD dwBytes1; |
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312 LPVOID lpvPtr2; |
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313 DWORD dwBytes2; |
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314 |
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315 // Lock the buffer |
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316 res = IDirectSoundBuffer_Lock(hdsbuf,write_offset, len, &lpvPtr1, &dwBytes1, &lpvPtr2, &dwBytes2, 0); |
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317 // If the buffer was lost, restore and retry lock. |
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318 if (DSERR_BUFFERLOST == res) |
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319 { |
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320 IDirectSoundBuffer_Restore(hdsbuf); |
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321 res = IDirectSoundBuffer_Lock(hdsbuf,write_offset, len, &lpvPtr1, &dwBytes1, &lpvPtr2, &dwBytes2, 0); |
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322 } |
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323 |
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324 |
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325 if (SUCCEEDED(res)) |
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326 { |
14245 | 327 if( (ao_data.channels == 6) && (ao_data.format!=AF_FORMAT_AC3) ) { |
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328 // reorder channels while writing to pointers. |
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329 // it's this easy because buffer size and len are always |
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330 // aligned to multiples of channels*bytespersample |
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331 // there's probably some room for speed improvements here |
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332 const int chantable[6] = {0, 1, 4, 5, 2, 3}; // reorder "matrix" |
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333 int i, j; |
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334 int numsamp,sampsize; |
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335 |
14249 | 336 sampsize = af_fmt2bits(ao_data.format)>>3; // bytes per sample |
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337 numsamp = dwBytes1 / (ao_data.channels * sampsize); // number of samples for each channel in this buffer |
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338 |
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339 for( i = 0; i < numsamp; i++ ) for( j = 0; j < ao_data.channels; j++ ) { |
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340 memcpy(lpvPtr1+(i*ao_data.channels*sampsize)+(chantable[j]*sampsize),data+(i*ao_data.channels*sampsize)+(j*sampsize),sampsize); |
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341 } |
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342 |
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343 if (NULL != lpvPtr2 ) |
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344 { |
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345 numsamp = dwBytes2 / (ao_data.channels * sampsize); |
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346 for( i = 0; i < numsamp; i++ ) for( j = 0; j < ao_data.channels; j++ ) { |
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347 memcpy(lpvPtr2+(i*ao_data.channels*sampsize)+(chantable[j]*sampsize),data+dwBytes1+(i*ao_data.channels*sampsize)+(j*sampsize),sampsize); |
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348 } |
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349 } |
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350 |
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351 write_offset+=dwBytes1+dwBytes2; |
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352 if(write_offset>=buffer_size)write_offset=dwBytes2; |
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353 } else { |
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354 // Write to pointers without reordering. |
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355 memcpy(lpvPtr1,data,dwBytes1); |
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356 if (NULL != lpvPtr2 )memcpy(lpvPtr2,data+dwBytes1,dwBytes2); |
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357 write_offset+=dwBytes1+dwBytes2; |
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358 if(write_offset>=buffer_size)write_offset=dwBytes2; |
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359 } |
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360 |
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361 // Release the data back to DirectSound. |
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362 res = IDirectSoundBuffer_Unlock(hdsbuf,lpvPtr1,dwBytes1,lpvPtr2,dwBytes2); |
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363 if (SUCCEEDED(res)) |
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364 { |
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365 // Success. |
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366 DWORD status; |
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367 IDirectSoundBuffer_GetStatus(hdsbuf, &status); |
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368 if (!(status & DSBSTATUS_PLAYING)){ |
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369 res = IDirectSoundBuffer_Play(hdsbuf, 0, 0, DSBPLAY_LOOPING); |
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370 } |
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371 return dwBytes1+dwBytes2; |
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372 } |
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373 } |
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374 // Lock, Unlock, or Restore failed. |
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375 return 0; |
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376 } |
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377 |
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378 /***************************************************************************************/ |
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379 |
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380 /** |
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381 \brief handle control commands |
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382 \param cmd command |
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383 \param arg argument |
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384 \return CONTROL_OK or -1 in case the command can't be handled |
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385 */ |
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386 static int control(int cmd, void *arg) |
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387 { |
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388 DWORD volume; |
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389 switch (cmd) { |
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390 case AOCONTROL_GET_VOLUME: { |
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391 ao_control_vol_t* vol = (ao_control_vol_t*)arg; |
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392 IDirectSoundBuffer_GetVolume(hdsbuf, &volume); |
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393 vol->left = vol->right = pow(10.0, (float)(volume+10000) / 5000.0); |
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394 //printf("ao_dsound: volume: %f\n",vol->left); |
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395 return CONTROL_OK; |
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396 } |
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397 case AOCONTROL_SET_VOLUME: { |
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398 ao_control_vol_t* vol = (ao_control_vol_t*)arg; |
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399 volume = (DWORD)(log10(vol->right) * 5000.0) - 10000; |
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400 IDirectSoundBuffer_SetVolume(hdsbuf, volume); |
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401 //printf("ao_dsound: volume: %f\n",vol->left); |
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402 return CONTROL_OK; |
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403 } |
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404 } |
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405 return -1; |
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406 } |
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407 |
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408 /** |
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409 \brief setup sound device |
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410 \param rate samplerate |
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411 \param channels number of channels |
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412 \param format format |
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413 \param flags unused |
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414 \return 1=success 0=fail |
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415 */ |
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416 static int init(int rate, int channels, int format, int flags) |
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417 { |
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418 int res; |
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419 if (!InitDirectSound()) return 0; |
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420 |
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421 // ok, now create the buffers |
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422 WAVEFORMATEXTENSIBLE wformat; |
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423 DSBUFFERDESC dsbpridesc; |
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424 DSBUFFERDESC dsbdesc; |
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425 |
14135 | 426 //check if the format is supported in general |
427 switch(format){ | |
14245 | 428 case AF_FORMAT_AC3: |
429 case AF_FORMAT_S24_LE: | |
430 case AF_FORMAT_S16_LE: | |
431 case AF_FORMAT_S8: | |
14135 | 432 break; |
433 default: | |
14264 | 434 mp_msg(MSGT_AO, MSGL_V,"ao_dsound: format %s not supported defaulting to Signed 16-bit Little-Endian\n",af_fmt2str_short(format)); |
14245 | 435 format=AF_FORMAT_S16_LE; |
14135 | 436 } |
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437 //fill global ao_data |
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438 ao_data.channels = channels; |
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439 ao_data.samplerate = rate; |
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440 ao_data.format = format; |
14249 | 441 ao_data.bps = channels * rate * (af_fmt2bits(format)>>3); |
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442 if(ao_data.buffersize==-1) ao_data.buffersize = ao_data.bps; // space for 1 sec |
14264 | 443 mp_msg(MSGT_AO, MSGL_V,"ao_dsound: Samplerate:%iHz Channels:%i Format:%s\n", rate, channels, af_fmt2str_short(format)); |
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444 mp_msg(MSGT_AO, MSGL_V,"ao_dsound: Buffersize:%d bytes (%d msec)\n", ao_data.buffersize, ao_data.buffersize / ao_data.bps * 1000); |
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445 |
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446 //fill waveformatex |
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447 ZeroMemory(&wformat, sizeof(WAVEFORMATEXTENSIBLE)); |
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448 wformat.Format.cbSize = (channels > 2) ? sizeof(WAVEFORMATEXTENSIBLE)-sizeof(WAVEFORMATEX) : 0; |
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449 wformat.Format.nChannels = channels; |
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450 wformat.Format.nSamplesPerSec = rate; |
14245 | 451 if (format == AF_FORMAT_AC3) { |
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452 wformat.Format.wFormatTag = WAVE_FORMAT_DOLBY_AC3_SPDIF; |
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453 wformat.Format.wBitsPerSample = 16; |
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454 wformat.Format.nBlockAlign = 4; |
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455 } else { |
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456 wformat.Format.wFormatTag = (channels > 2) ? WAVE_FORMAT_EXTENSIBLE : WAVE_FORMAT_PCM; |
14249 | 457 wformat.Format.wBitsPerSample = af_fmt2bits(format); |
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458 wformat.Format.nBlockAlign = wformat.Format.nChannels * (wformat.Format.wBitsPerSample >> 3); |
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459 } |
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460 |
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461 // fill in primary sound buffer descriptor |
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462 memset(&dsbpridesc, 0, sizeof(DSBUFFERDESC)); |
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463 dsbpridesc.dwSize = sizeof(DSBUFFERDESC); |
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464 dsbpridesc.dwFlags = DSBCAPS_PRIMARYBUFFER; |
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465 dsbpridesc.dwBufferBytes = 0; |
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466 dsbpridesc.lpwfxFormat = NULL; |
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467 |
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468 |
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469 // fill in the secondary sound buffer (=stream buffer) descriptor |
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470 memset(&dsbdesc, 0, sizeof(DSBUFFERDESC)); |
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471 dsbdesc.dwSize = sizeof(DSBUFFERDESC); |
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472 dsbdesc.dwFlags = DSBCAPS_GETCURRENTPOSITION2 /** Better position accuracy */ |
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473 | DSBCAPS_GLOBALFOCUS /** Allows background playing */ |
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474 | DSBCAPS_CTRLVOLUME; /** volume control enabled */ |
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475 |
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476 if (channels > 2) { |
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477 wformat.dwChannelMask = channel_mask[channels - 3]; |
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478 wformat.SubFormat = KSDATAFORMAT_SUBTYPE_PCM; |
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479 wformat.Samples.wValidBitsPerSample = wformat.Format.wBitsPerSample; |
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480 // Needed for 5.1 on emu101k - shit soundblaster |
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481 dsbdesc.dwFlags |= DSBCAPS_LOCHARDWARE; |
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482 } |
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483 wformat.Format.nAvgBytesPerSec = wformat.Format.nSamplesPerSec * wformat.Format.nBlockAlign; |
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484 |
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485 dsbdesc.dwBufferBytes = ao_data.buffersize; |
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486 dsbdesc.lpwfxFormat = (WAVEFORMATEX *)&wformat; |
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487 buffer_size = dsbdesc.dwBufferBytes; |
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488 write_offset = 0; |
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489 min_free_space = wformat.Format.nBlockAlign; |
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490 ao_data.outburst = wformat.Format.nBlockAlign * 512; |
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491 |
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492 // create primary buffer and set its format |
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493 |
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494 res = IDirectSound_CreateSoundBuffer( hds, &dsbpridesc, &hdspribuf, NULL ); |
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495 if ( res != DS_OK ) { |
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496 UninitDirectSound(); |
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497 mp_msg(MSGT_AO, MSGL_ERR,"ao_dsound: cannot create primary buffer (%s)\n", dserr2str(res)); |
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498 return 0; |
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499 } |
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500 res = IDirectSoundBuffer_SetFormat( hdspribuf, (WAVEFORMATEX *)&wformat ); |
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501 if ( res != DS_OK ) mp_msg(MSGT_AO, MSGL_WARN,"ao_dsound: cannot set primary buffer format (%s), using standard setting (bad quality)", dserr2str(res)); |
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502 |
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503 mp_msg(MSGT_AO, MSGL_V, "ao_dsound: primary buffer created\n"); |
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504 |
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505 // now create the stream buffer |
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506 |
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507 res = IDirectSound_CreateSoundBuffer(hds, &dsbdesc, &hdsbuf, NULL); |
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508 if (res != DS_OK) { |
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509 if (dsbdesc.dwFlags & DSBCAPS_LOCHARDWARE) { |
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510 // Try without DSBCAPS_LOCHARDWARE |
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511 dsbdesc.dwFlags &= ~DSBCAPS_LOCHARDWARE; |
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512 res = IDirectSound_CreateSoundBuffer(hds, &dsbdesc, &hdsbuf, NULL); |
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513 } |
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directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
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514 if (res != DS_OK) { |
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directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
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|
515 UninitDirectSound(); |
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516 mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: cannot create secondary (stream)buffer (%s)\n", dserr2str(res)); |
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directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
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517 return 0; |
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518 } |
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519 } |
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520 mp_msg(MSGT_AO, MSGL_V, "ao_dsound: secondary (stream)buffer created\n"); |
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521 return 1; |
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522 } |
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523 |
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directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
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524 |
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directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
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525 |
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526 /** |
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directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
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527 \brief stop playing and empty buffers (for seeking/pause) |
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directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
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528 */ |
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529 static void reset(void) |
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530 { |
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531 IDirectSoundBuffer_Stop(hdsbuf); |
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532 // reset directsound buffer |
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533 IDirectSoundBuffer_SetCurrentPosition(hdsbuf, 0); |
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directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
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534 write_offset=0; |
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|
535 } |
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536 |
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537 /** |
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538 \brief stop playing, keep buffers (for pause) |
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539 */ |
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540 static void audio_pause(void) |
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541 { |
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directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
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542 IDirectSoundBuffer_Stop(hdsbuf); |
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543 } |
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544 |
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directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
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|
545 /** |
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directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
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546 \brief resume playing, after audio_pause() |
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547 */ |
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548 static void audio_resume(void) |
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549 { |
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550 IDirectSoundBuffer_Play(hdsbuf, 0, 0, DSBPLAY_LOOPING); |
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551 } |
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directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
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552 |
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directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
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553 /** |
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directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
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554 \brief close audio device |
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support immed flag, always initialize write_offset, min_free_space doesn't seem to be required anymore after Florian Dietrichs patches
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555 \param immed stop playback immediately |
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556 */ |
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557 static void uninit(int immed) |
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directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
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558 { |
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559 if(immed)reset(); |
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560 else{ |
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561 DWORD status; |
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562 IDirectSoundBuffer_Play(hdsbuf, 0, 0, 0); |
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563 while(!IDirectSoundBuffer_GetStatus(hdsbuf,&status) && (status&DSBSTATUS_PLAYING)) |
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564 usec_sleep(20000); |
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565 } |
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directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
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|
566 DestroyBuffer(); |
70d8f1975fc8
directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
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567 UninitDirectSound(); |
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568 } |
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|
569 |
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directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
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|
570 /** |
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directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
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571 \brief find out how many bytes can be written into the audio buffer without |
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directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
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572 \return free space in bytes, has to return 0 if the buffer is almost full |
70d8f1975fc8
directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
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573 */ |
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|
574 static int get_space(void) |
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575 { |
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directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
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|
576 int space; |
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577 DWORD play_offset; |
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|
578 IDirectSoundBuffer_GetCurrentPosition(hdsbuf,&play_offset,NULL); |
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579 space=buffer_size-(write_offset-play_offset); |
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|
580 // | | <-- const --> | | | |
70d8f1975fc8
directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
faust3
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|
581 // buffer start play_cursor write_cursor write_offset buffer end |
70d8f1975fc8
directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
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|
582 // play_cursor is the actual postion of the play cursor |
70d8f1975fc8
directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
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583 // write_cursor is the position after which it is assumed to be save to write data |
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directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
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|
584 // write_offset is the postion where we actually write the data to |
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directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
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585 if(space > buffer_size)space -= buffer_size; // write_offset < play_offset |
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directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
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|
586 if(space < min_free_space)return 0; |
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14555
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587 return space-min_free_space; |
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|
588 } |
70d8f1975fc8
directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
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|
589 |
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directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
faust3
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|
590 /** |
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directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
faust3
parents:
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|
591 \brief play 'len' bytes of 'data' |
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directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
faust3
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|
592 \param data pointer to the data to play |
70d8f1975fc8
directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
faust3
parents:
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|
593 \param len size in bytes of the data buffer, gets rounded down to outburst*n |
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directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
faust3
parents:
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|
594 \param flags currently unused |
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directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
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|
595 \return number of played bytes |
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directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
faust3
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|
596 */ |
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directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
faust3
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diff
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|
597 static int play(void* data, int len, int flags) |
70d8f1975fc8
directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
faust3
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|
598 { |
13674
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faust3
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13640
diff
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|
599 DWORD play_offset; |
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channel reorder patch by Florian Dietrich <flodt8 at yahoo.de>
faust3
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|
600 int space; |
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channel reorder patch by Florian Dietrich <flodt8 at yahoo.de>
faust3
parents:
13640
diff
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|
601 |
b7322244e53c
channel reorder patch by Florian Dietrich <flodt8 at yahoo.de>
faust3
parents:
13640
diff
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|
602 // make sure we have enough space to write data |
b7322244e53c
channel reorder patch by Florian Dietrich <flodt8 at yahoo.de>
faust3
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13640
diff
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|
603 IDirectSoundBuffer_GetCurrentPosition(hdsbuf,&play_offset,NULL); |
b7322244e53c
channel reorder patch by Florian Dietrich <flodt8 at yahoo.de>
faust3
parents:
13640
diff
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|
604 space=buffer_size-(write_offset-play_offset); |
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channel reorder patch by Florian Dietrich <flodt8 at yahoo.de>
faust3
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13640
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|
605 if(space > buffer_size)space -= buffer_size; // write_offset < play_offset |
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faust3
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13640
diff
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|
606 if(space < len) len = space; |
b7322244e53c
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faust3
parents:
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diff
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|
607 |
20250
7cfd3a04d537
Implement AOPLAY_FINAL_CHUNK support for dshow and win32 out.
reimar
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19614
diff
changeset
|
608 if (!(flags & AOPLAY_FINAL_CHUNK)) |
13460
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directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
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609 len = (len / ao_data.outburst) * ao_data.outburst; |
70d8f1975fc8
directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
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|
610 return write_buffer(data, len); |
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directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
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|
611 } |
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directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
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|
612 |
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|
613 /** |
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directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
faust3
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|
614 \brief get the delay between the first and last sample in the buffer |
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directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
faust3
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|
615 \return delay in seconds |
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directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
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616 */ |
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617 static float get_delay(void) |
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directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
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|
618 { |
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directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
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|
619 DWORD play_offset; |
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|
620 int space; |
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|
621 IDirectSoundBuffer_GetCurrentPosition(hdsbuf,&play_offset,NULL); |
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|
622 space=play_offset-write_offset; |
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|
623 if(space <= 0)space += buffer_size; |
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directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
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624 return (float)(buffer_size - space) / (float)ao_data.bps; |
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directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
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|
625 } |