Mercurial > mplayer.hg
annotate libao2/ao_dsound.c @ 14585:81059277538c
Be less verbose with -v (do not print a line for each demuxed packet)
author | rtognimp |
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date | Mon, 24 Jan 2005 21:52:24 +0000 |
parents | 6fb9eca97d41 |
children | 2f4f347bd5e4 |
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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 |
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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 \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 |
14479 | 31 #include "config.h" |
14245 | 32 #include "libaf/af_format.h" |
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33 #include "audio_out.h" |
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34 #include "audio_out_internal.h" |
14123 | 35 #include "mp_msg.h" |
36 #include "libvo/fastmemcpy.h" | |
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37 #include "osdep/timer.h" |
14398 | 38 #include "subopt-helper.h" |
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39 |
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40 |
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41 static ao_info_t info = |
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42 { |
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43 "Windows DirectSound audio output", |
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44 "dsound", |
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45 "Gabor Szecsi <deje@miki.hu>", |
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46 "" |
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47 }; |
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48 |
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49 LIBAO_EXTERN(dsound) |
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50 |
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51 /** |
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52 \todo use the definitions from the win32 api headers when they define these |
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53 */ |
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54 #if 1 |
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55 #define WAVE_FORMAT_IEEE_FLOAT 0x0003 |
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56 #define WAVE_FORMAT_DOLBY_AC3_SPDIF 0x0092 |
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57 #define WAVE_FORMAT_EXTENSIBLE 0xFFFE |
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58 |
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59 static const GUID KSDATAFORMAT_SUBTYPE_PCM = {0x1,0x0000,0x0010, {0x80,0x00,0x00,0xaa,0x00,0x38,0x9b,0x71}}; |
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60 |
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61 #define SPEAKER_FRONT_LEFT 0x1 |
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62 #define SPEAKER_FRONT_RIGHT 0x2 |
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63 #define SPEAKER_FRONT_CENTER 0x4 |
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64 #define SPEAKER_LOW_FREQUENCY 0x8 |
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65 #define SPEAKER_BACK_LEFT 0x10 |
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66 #define SPEAKER_BACK_RIGHT 0x20 |
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67 #define SPEAKER_FRONT_LEFT_OF_CENTER 0x40 |
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68 #define SPEAKER_FRONT_RIGHT_OF_CENTER 0x80 |
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69 #define SPEAKER_BACK_CENTER 0x100 |
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70 #define SPEAKER_SIDE_LEFT 0x200 |
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71 #define SPEAKER_SIDE_RIGHT 0x400 |
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72 #define SPEAKER_TOP_CENTER 0x800 |
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73 #define SPEAKER_TOP_FRONT_LEFT 0x1000 |
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74 #define SPEAKER_TOP_FRONT_CENTER 0x2000 |
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75 #define SPEAKER_TOP_FRONT_RIGHT 0x4000 |
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76 #define SPEAKER_TOP_BACK_LEFT 0x8000 |
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77 #define SPEAKER_TOP_BACK_CENTER 0x10000 |
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78 #define SPEAKER_TOP_BACK_RIGHT 0x20000 |
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79 #define SPEAKER_RESERVED 0x80000000 |
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80 |
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81 #define DSSPEAKER_HEADPHONE 0x00000001 |
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82 #define DSSPEAKER_MONO 0x00000002 |
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83 #define DSSPEAKER_QUAD 0x00000003 |
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84 #define DSSPEAKER_STEREO 0x00000004 |
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85 #define DSSPEAKER_SURROUND 0x00000005 |
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86 #define DSSPEAKER_5POINT1 0x00000006 |
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87 |
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88 #ifndef _WAVEFORMATEXTENSIBLE_ |
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89 typedef struct { |
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90 WAVEFORMATEX Format; |
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91 union { |
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92 WORD wValidBitsPerSample; /* bits of precision */ |
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93 WORD wSamplesPerBlock; /* valid if wBitsPerSample==0 */ |
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94 WORD wReserved; /* If neither applies, set to zero. */ |
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95 } Samples; |
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96 DWORD dwChannelMask; /* which channels are */ |
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97 /* present in stream */ |
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98 GUID SubFormat; |
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99 } WAVEFORMATEXTENSIBLE, *PWAVEFORMATEXTENSIBLE; |
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100 #endif |
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101 |
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102 #endif |
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103 |
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104 static const int channel_mask[] = { |
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105 SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_LOW_FREQUENCY, |
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106 SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT, |
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107 SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | SPEAKER_LOW_FREQUENCY, |
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108 SPEAKER_FRONT_LEFT | SPEAKER_FRONT_CENTER | SPEAKER_FRONT_RIGHT | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | SPEAKER_LOW_FREQUENCY |
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109 }; |
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110 |
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111 static HINSTANCE hdsound_dll = NULL; ///handle to the dll |
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112 static LPDIRECTSOUND hds = NULL; ///direct sound object |
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113 static LPDIRECTSOUNDBUFFER hdspribuf = NULL; ///primary direct sound buffer |
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114 static LPDIRECTSOUNDBUFFER hdsbuf = NULL; ///secondary direct sound buffer (stream buffer) |
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115 static int buffer_size = 0; ///size in bytes of the direct sound buffer |
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116 static int write_offset = 0; ///offset of the write cursor in the direct sound buffer |
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117 static int min_free_space = 0; ///if the free space is below this value get_space() will return 0 |
14398 | 118 static int device_num = 0; ///wanted device number |
119 static GUID device; ///guid of the device | |
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120 |
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121 /***************************************************************************************/ |
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122 |
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123 /** |
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124 \brief output error message |
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125 \param err error code |
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126 \return string with the error message |
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127 */ |
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128 static char * dserr2str(int err) |
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129 { |
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130 switch (err) { |
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131 case DS_OK: return "DS_OK"; |
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132 case DS_NO_VIRTUALIZATION: return "DS_NO_VIRTUALIZATION"; |
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133 case DSERR_ALLOCATED: return "DS_NO_VIRTUALIZATION"; |
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134 case DSERR_CONTROLUNAVAIL: return "DSERR_CONTROLUNAVAIL"; |
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135 case DSERR_INVALIDPARAM: return "DSERR_INVALIDPARAM"; |
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136 case DSERR_INVALIDCALL: return "DSERR_INVALIDCALL"; |
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137 case DSERR_GENERIC: return "DSERR_GENERIC"; |
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138 case DSERR_PRIOLEVELNEEDED: return "DSERR_PRIOLEVELNEEDED"; |
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139 case DSERR_OUTOFMEMORY: return "DSERR_OUTOFMEMORY"; |
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140 case DSERR_BADFORMAT: return "DSERR_BADFORMAT"; |
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141 case DSERR_UNSUPPORTED: return "DSERR_UNSUPPORTED"; |
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142 case DSERR_NODRIVER: return "DSERR_NODRIVER"; |
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143 case DSERR_ALREADYINITIALIZED: return "DSERR_ALREADYINITIALIZED"; |
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144 case DSERR_NOAGGREGATION: return "DSERR_NOAGGREGATION"; |
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145 case DSERR_BUFFERLOST: return "DSERR_BUFFERLOST"; |
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146 case DSERR_OTHERAPPHASPRIO: return "DSERR_OTHERAPPHASPRIO"; |
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147 case DSERR_UNINITIALIZED: return "DSERR_UNINITIALIZED"; |
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148 case DSERR_NOINTERFACE: return "DSERR_NOINTERFACE"; |
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149 case DSERR_ACCESSDENIED: return "DSERR_ACCESSDENIED"; |
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150 default: return "unknown"; |
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151 } |
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152 } |
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153 |
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154 /** |
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155 \brief uninitialize direct sound |
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156 */ |
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157 static void UninitDirectSound(void) |
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158 { |
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159 // finally release the DirectSound object |
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160 if (hds) { |
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161 IDirectSound_Release(hds); |
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162 hds = NULL; |
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163 } |
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164 // free DSOUND.DLL |
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165 if (hdsound_dll) { |
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166 FreeLibrary(hdsound_dll); |
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167 hdsound_dll = NULL; |
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168 } |
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169 mp_msg(MSGT_AO, MSGL_V, "ao_dsound: DirectSound uninitialized\n"); |
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170 } |
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171 |
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172 /** |
14398 | 173 \brief print the commandline help |
174 */ | |
175 static void print_help() | |
176 { | |
177 mp_msg(MSGT_AO, MSGL_FATAL, | |
178 "\n-ao dsound commandline help:\n" | |
179 "Example: mplayer -ao dsound:device=1\n" | |
180 " sets 1st device\n" | |
181 "\nOptions:\n" | |
182 " device=<device-number>\n" | |
183 " Sets device number, use -v to get a list\n"); | |
184 } | |
185 | |
186 | |
187 /** | |
188 \brief enumerate direct sound devices | |
189 \return TRUE to continue with the enumeration | |
190 */ | |
191 static BOOL CALLBACK DirectSoundEnum(LPGUID guid,LPCSTR desc,LPCSTR module,LPVOID context) | |
192 { | |
193 int* device_index=context; | |
194 mp_msg(MSGT_AO, MSGL_V,"%i %s ",*device_index,desc); | |
195 if(device_num==*device_index){ | |
196 mp_msg(MSGT_AO, MSGL_V,"<--"); | |
197 if(guid){ | |
198 memcpy(&device,guid,sizeof(GUID)); | |
199 } | |
200 } | |
201 mp_msg(MSGT_AO, MSGL_V,"\n"); | |
202 (*device_index)++; | |
203 return TRUE; | |
204 } | |
205 | |
206 | |
207 /** | |
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208 \brief initilize direct sound |
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209 \return 0 if error, 1 if ok |
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210 */ |
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211 static int InitDirectSound(void) |
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212 { |
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213 DSCAPS dscaps; |
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214 |
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215 // initialize directsound |
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216 HRESULT (WINAPI *OurDirectSoundCreate)(LPGUID, LPDIRECTSOUND *, LPUNKNOWN); |
14398 | 217 HRESULT (WINAPI *OurDirectSoundEnumerate)(LPDSENUMCALLBACKA, LPVOID); |
218 int device_index=0; | |
219 opt_t subopts[] = { | |
220 {"device", OPT_ARG_INT, &device_num,NULL}, | |
221 {NULL} | |
222 }; | |
223 if (subopt_parse(ao_subdevice, subopts) != 0) { | |
224 print_help(); | |
225 return 0; | |
226 } | |
227 | |
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228 hdsound_dll = LoadLibrary("DSOUND.DLL"); |
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229 if (hdsound_dll == NULL) { |
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230 mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: cannot load DSOUND.DLL\n"); |
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231 return 0; |
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232 } |
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233 OurDirectSoundCreate = (void*)GetProcAddress(hdsound_dll, "DirectSoundCreate"); |
14398 | 234 OurDirectSoundEnumerate = (void*)GetProcAddress(hdsound_dll, "DirectSoundEnumerateA"); |
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235 |
14398 | 236 if (OurDirectSoundCreate == NULL || OurDirectSoundEnumerate == NULL) { |
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237 mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: GetProcAddress FAILED\n"); |
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238 FreeLibrary(hdsound_dll); |
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239 return 0; |
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240 } |
14398 | 241 |
242 // Enumerate all directsound devices | |
243 mp_msg(MSGT_AO, MSGL_V,"ao_dsound: Output Devices:\n"); | |
244 OurDirectSoundEnumerate(DirectSoundEnum,&device_index); | |
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245 |
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246 // Create the direct sound object |
14398 | 247 if FAILED(OurDirectSoundCreate((device_num)?&device:NULL, &hds, NULL )) { |
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248 mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: cannot create a DirectSound device\n"); |
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249 FreeLibrary(hdsound_dll); |
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250 return 0; |
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251 } |
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252 |
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253 /* Set DirectSound Cooperative level, ie what control we want over Windows |
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254 * sound device. In our case, DSSCL_EXCLUSIVE means that we can modify the |
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255 * settings of the primary buffer, but also that only the sound of our |
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256 * application will be hearable when it will have the focus. |
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257 * !!! (this is not really working as intended yet because to set the |
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258 * cooperative level you need the window handle of your application, and |
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259 * I don't know of any easy way to get it. Especially since we might play |
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260 * sound without any video, and so what window handle should we use ??? |
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261 * The hack for now is to use the Desktop window handle - it seems to be |
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262 * working */ |
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263 if (IDirectSound_SetCooperativeLevel(hds, GetDesktopWindow(), DSSCL_EXCLUSIVE)) { |
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264 mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: cannot set direct sound cooperative level\n"); |
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265 IDirectSound_Release(hds); |
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266 FreeLibrary(hdsound_dll); |
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267 return 0; |
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268 } |
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269 mp_msg(MSGT_AO, MSGL_V, "ao_dsound: DirectSound initialized\n"); |
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270 |
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271 memset(&dscaps, 0, sizeof(DSCAPS)); |
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272 dscaps.dwSize = sizeof(DSCAPS); |
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273 if (DS_OK == IDirectSound_GetCaps(hds, &dscaps)) { |
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274 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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275 } else { |
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276 mp_msg(MSGT_AO, MSGL_V, "ao_dsound: cannot get device capabilities\n"); |
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277 } |
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278 |
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279 return 1; |
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280 } |
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281 |
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282 /** |
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283 \brief destroy the direct sound buffer |
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284 */ |
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285 static void DestroyBuffer(void) |
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286 { |
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287 if (hdsbuf) { |
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288 IDirectSoundBuffer_Release(hdsbuf); |
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289 hdsbuf = NULL; |
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290 } |
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291 if (hdspribuf) { |
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292 IDirectSoundBuffer_Release(hdspribuf); |
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293 hdspribuf = NULL; |
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294 } |
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295 } |
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296 |
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297 /** |
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298 \brief fill sound buffer |
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299 \param data pointer to the sound data to copy |
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300 \param len length of the data to copy in bytes |
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301 \return number of copyed bytes |
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302 */ |
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303 static int write_buffer(unsigned char *data, int len) |
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304 { |
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305 HRESULT res; |
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306 LPVOID lpvPtr1; |
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307 DWORD dwBytes1; |
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308 LPVOID lpvPtr2; |
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309 DWORD dwBytes2; |
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310 |
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311 // Lock the buffer |
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312 res = IDirectSoundBuffer_Lock(hdsbuf,write_offset, len, &lpvPtr1, &dwBytes1, &lpvPtr2, &dwBytes2, 0); |
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313 // If the buffer was lost, restore and retry lock. |
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314 if (DSERR_BUFFERLOST == res) |
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315 { |
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316 IDirectSoundBuffer_Restore(hdsbuf); |
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317 res = IDirectSoundBuffer_Lock(hdsbuf,write_offset, len, &lpvPtr1, &dwBytes1, &lpvPtr2, &dwBytes2, 0); |
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318 } |
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319 |
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320 |
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321 if (SUCCEEDED(res)) |
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322 { |
14245 | 323 if( (ao_data.channels == 6) && (ao_data.format!=AF_FORMAT_AC3) ) { |
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324 // reorder channels while writing to pointers. |
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325 // it's this easy because buffer size and len are always |
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326 // aligned to multiples of channels*bytespersample |
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327 // there's probably some room for speed improvements here |
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328 const int chantable[6] = {0, 1, 4, 5, 2, 3}; // reorder "matrix" |
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329 int i, j; |
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330 int numsamp,sampsize; |
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331 |
14249 | 332 sampsize = af_fmt2bits(ao_data.format)>>3; // bytes per sample |
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333 numsamp = dwBytes1 / (ao_data.channels * sampsize); // number of samples for each channel in this buffer |
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334 |
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335 for( i = 0; i < numsamp; i++ ) for( j = 0; j < ao_data.channels; j++ ) { |
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336 memcpy(lpvPtr1+(i*ao_data.channels*sampsize)+(chantable[j]*sampsize),data+(i*ao_data.channels*sampsize)+(j*sampsize),sampsize); |
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337 } |
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338 |
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339 if (NULL != lpvPtr2 ) |
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340 { |
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341 numsamp = dwBytes2 / (ao_data.channels * sampsize); |
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342 for( i = 0; i < numsamp; i++ ) for( j = 0; j < ao_data.channels; j++ ) { |
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343 memcpy(lpvPtr2+(i*ao_data.channels*sampsize)+(chantable[j]*sampsize),data+dwBytes1+(i*ao_data.channels*sampsize)+(j*sampsize),sampsize); |
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344 } |
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345 } |
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346 |
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347 write_offset+=dwBytes1+dwBytes2; |
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348 if(write_offset>=buffer_size)write_offset=dwBytes2; |
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349 } else { |
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350 // Write to pointers without reordering. |
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351 memcpy(lpvPtr1,data,dwBytes1); |
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352 if (NULL != lpvPtr2 )memcpy(lpvPtr2,data+dwBytes1,dwBytes2); |
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353 write_offset+=dwBytes1+dwBytes2; |
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354 if(write_offset>=buffer_size)write_offset=dwBytes2; |
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355 } |
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356 |
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357 // Release the data back to DirectSound. |
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358 res = IDirectSoundBuffer_Unlock(hdsbuf,lpvPtr1,dwBytes1,lpvPtr2,dwBytes2); |
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359 if (SUCCEEDED(res)) |
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360 { |
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361 // Success. |
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362 DWORD status; |
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363 IDirectSoundBuffer_GetStatus(hdsbuf, &status); |
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364 if (!(status & DSBSTATUS_PLAYING)){ |
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365 res = IDirectSoundBuffer_Play(hdsbuf, 0, 0, DSBPLAY_LOOPING); |
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366 } |
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367 return dwBytes1+dwBytes2; |
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368 } |
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369 } |
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370 // Lock, Unlock, or Restore failed. |
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371 return 0; |
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372 } |
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373 |
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374 /***************************************************************************************/ |
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375 |
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376 /** |
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377 \brief handle control commands |
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378 \param cmd command |
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379 \param arg argument |
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380 \return CONTROL_OK or -1 in case the command can't be handled |
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381 */ |
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382 static int control(int cmd, void *arg) |
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383 { |
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384 DWORD volume; |
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385 switch (cmd) { |
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386 case AOCONTROL_GET_VOLUME: { |
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387 ao_control_vol_t* vol = (ao_control_vol_t*)arg; |
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388 IDirectSoundBuffer_GetVolume(hdsbuf, &volume); |
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389 vol->left = vol->right = (float)(volume+10000) / 100.0; |
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390 //printf("ao_dsound: volume: %f\n",vol->left); |
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391 return CONTROL_OK; |
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392 } |
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393 case AOCONTROL_SET_VOLUME: { |
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394 ao_control_vol_t* vol = (ao_control_vol_t*)arg; |
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395 volume = (vol->right * 100.0)-10000; |
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396 IDirectSoundBuffer_SetVolume(hdsbuf, volume); |
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397 //printf("ao_dsound: volume: %f\n",vol->left); |
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398 return CONTROL_OK; |
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399 } |
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400 } |
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401 return -1; |
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402 } |
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403 |
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404 /** |
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405 \brief setup sound device |
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406 \param rate samplerate |
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407 \param channels number of channels |
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408 \param format format |
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409 \param flags unused |
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410 \return 1=success 0=fail |
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411 */ |
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412 static int init(int rate, int channels, int format, int flags) |
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413 { |
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414 int res; |
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415 if (!InitDirectSound()) return 0; |
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416 |
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417 // ok, now create the buffers |
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418 WAVEFORMATEXTENSIBLE wformat; |
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419 DSBUFFERDESC dsbpridesc; |
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420 DSBUFFERDESC dsbdesc; |
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421 |
14135 | 422 //check if the format is supported in general |
423 switch(format){ | |
14245 | 424 case AF_FORMAT_AC3: |
425 case AF_FORMAT_S24_LE: | |
426 case AF_FORMAT_S16_LE: | |
427 case AF_FORMAT_S8: | |
14135 | 428 break; |
429 default: | |
14264 | 430 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 | 431 format=AF_FORMAT_S16_LE; |
14135 | 432 } |
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433 //fill global ao_data |
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434 ao_data.channels = channels; |
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435 ao_data.samplerate = rate; |
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436 ao_data.format = format; |
14249 | 437 ao_data.bps = channels * rate * (af_fmt2bits(format)>>3); |
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438 if(ao_data.buffersize==-1) ao_data.buffersize = ao_data.bps; // space for 1 sec |
14264 | 439 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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440 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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441 |
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442 //fill waveformatex |
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443 ZeroMemory(&wformat, sizeof(WAVEFORMATEXTENSIBLE)); |
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444 wformat.Format.cbSize = (channels > 2) ? sizeof(WAVEFORMATEXTENSIBLE)-sizeof(WAVEFORMATEX) : 0; |
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445 wformat.Format.nChannels = channels; |
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446 wformat.Format.nSamplesPerSec = rate; |
14245 | 447 if (format == AF_FORMAT_AC3) { |
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448 wformat.Format.wFormatTag = WAVE_FORMAT_DOLBY_AC3_SPDIF; |
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449 wformat.Format.wBitsPerSample = 16; |
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450 wformat.Format.nBlockAlign = 4; |
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451 } else { |
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452 wformat.Format.wFormatTag = (channels > 2) ? WAVE_FORMAT_EXTENSIBLE : WAVE_FORMAT_PCM; |
14249 | 453 wformat.Format.wBitsPerSample = af_fmt2bits(format); |
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454 wformat.Format.nBlockAlign = wformat.Format.nChannels * (wformat.Format.wBitsPerSample >> 3); |
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455 } |
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456 |
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457 // fill in primary sound buffer descriptor |
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458 memset(&dsbpridesc, 0, sizeof(DSBUFFERDESC)); |
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459 dsbpridesc.dwSize = sizeof(DSBUFFERDESC); |
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460 dsbpridesc.dwFlags = DSBCAPS_PRIMARYBUFFER; |
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461 dsbpridesc.dwBufferBytes = 0; |
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462 dsbpridesc.lpwfxFormat = NULL; |
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463 |
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464 |
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465 // fill in the secondary sound buffer (=stream buffer) descriptor |
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466 memset(&dsbdesc, 0, sizeof(DSBUFFERDESC)); |
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467 dsbdesc.dwSize = sizeof(DSBUFFERDESC); |
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468 dsbdesc.dwFlags = DSBCAPS_GETCURRENTPOSITION2 /** Better position accuracy */ |
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469 | DSBCAPS_GLOBALFOCUS /** Allows background playing */ |
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470 | DSBCAPS_CTRLVOLUME; /** volume control enabled */ |
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471 |
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472 if (channels > 2) { |
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473 wformat.dwChannelMask = channel_mask[channels - 3]; |
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474 wformat.SubFormat = KSDATAFORMAT_SUBTYPE_PCM; |
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475 wformat.Samples.wValidBitsPerSample = wformat.Format.wBitsPerSample; |
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476 // Needed for 5.1 on emu101k - shit soundblaster |
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477 dsbdesc.dwFlags |= DSBCAPS_LOCHARDWARE; |
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478 } |
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479 wformat.Format.nAvgBytesPerSec = wformat.Format.nSamplesPerSec * wformat.Format.nBlockAlign; |
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480 |
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481 dsbdesc.dwBufferBytes = ao_data.buffersize; |
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482 dsbdesc.lpwfxFormat = (WAVEFORMATEX *)&wformat; |
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483 buffer_size = dsbdesc.dwBufferBytes; |
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484 write_offset = 0; |
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485 ao_data.outburst = wformat.Format.nBlockAlign * 512; |
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486 |
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487 // create primary buffer and set its format |
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488 |
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489 res = IDirectSound_CreateSoundBuffer( hds, &dsbpridesc, &hdspribuf, NULL ); |
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490 if ( res != DS_OK ) { |
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491 UninitDirectSound(); |
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492 mp_msg(MSGT_AO, MSGL_ERR,"ao_dsound: cannot create primary buffer (%s)\n", dserr2str(res)); |
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493 return 0; |
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494 } |
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495 res = IDirectSoundBuffer_SetFormat( hdspribuf, (WAVEFORMATEX *)&wformat ); |
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496 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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497 |
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498 mp_msg(MSGT_AO, MSGL_V, "ao_dsound: primary buffer created\n"); |
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499 |
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500 // now create the stream buffer |
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501 |
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502 res = IDirectSound_CreateSoundBuffer(hds, &dsbdesc, &hdsbuf, NULL); |
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503 if (res != DS_OK) { |
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504 if (dsbdesc.dwFlags & DSBCAPS_LOCHARDWARE) { |
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505 // Try without DSBCAPS_LOCHARDWARE |
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506 dsbdesc.dwFlags &= ~DSBCAPS_LOCHARDWARE; |
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507 res = IDirectSound_CreateSoundBuffer(hds, &dsbdesc, &hdsbuf, NULL); |
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508 } |
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509 if (res != DS_OK) { |
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510 UninitDirectSound(); |
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511 mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: cannot create secondary (stream)buffer (%s)\n", dserr2str(res)); |
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512 return 0; |
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513 } |
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514 } |
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515 mp_msg(MSGT_AO, MSGL_V, "ao_dsound: secondary (stream)buffer created\n"); |
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516 return 1; |
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517 } |
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518 |
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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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|
519 |
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520 |
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521 /** |
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522 \brief stop playing and empty buffers (for seeking/pause) |
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523 */ |
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524 static void reset() |
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525 { |
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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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526 IDirectSoundBuffer_Stop(hdsbuf); |
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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 // reset directsound buffer |
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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 IDirectSoundBuffer_SetCurrentPosition(hdsbuf, 0); |
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529 write_offset=0; |
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530 } |
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531 |
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532 /** |
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533 \brief stop playing, keep buffers (for 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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534 */ |
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535 static void audio_pause() |
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536 { |
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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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537 IDirectSoundBuffer_Stop(hdsbuf); |
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538 } |
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539 |
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540 /** |
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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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541 \brief resume playing, after audio_pause() |
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542 */ |
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543 static void audio_resume() |
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544 { |
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545 IDirectSoundBuffer_Play(hdsbuf, 0, 0, DSBPLAY_LOOPING); |
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546 } |
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547 |
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548 /** |
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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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549 \brief close audio device |
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550 \param immed stop playback immediately |
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551 */ |
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552 static void uninit(int immed) |
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553 { |
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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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554 if(immed)reset(); |
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555 else{ |
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556 DWORD status; |
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557 IDirectSoundBuffer_Play(hdsbuf, 0, 0, 0); |
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558 while(!IDirectSoundBuffer_GetStatus(hdsbuf,&status) && (status&DSBSTATUS_PLAYING)) |
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559 usec_sleep(20000); |
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560 } |
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561 DestroyBuffer(); |
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562 UninitDirectSound(); |
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563 } |
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564 |
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565 /** |
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566 \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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567 \return free space in bytes, has to return 0 if the buffer is almost full |
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568 */ |
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569 static int get_space() |
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570 { |
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571 int space; |
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572 DWORD play_offset; |
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573 IDirectSoundBuffer_GetCurrentPosition(hdsbuf,&play_offset,NULL); |
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574 space=buffer_size-(write_offset-play_offset); |
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575 // | | <-- const --> | | | |
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576 // buffer start play_cursor write_cursor write_offset buffer end |
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577 // play_cursor is the actual postion of the play cursor |
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578 // write_cursor is the position after which it is assumed to be save to write data |
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579 // write_offset is the postion where we actually write the data to |
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580 if(space > buffer_size)space -= buffer_size; // write_offset < play_offset |
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581 if(space < min_free_space)return 0; |
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582 return space; |
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|
583 } |
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|
584 |
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|
585 /** |
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|
586 \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
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587 \param data pointer to the data to play |
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588 \param len size in bytes of the data buffer, gets rounded down to outburst*n |
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|
589 \param flags currently unused |
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590 \return number of played bytes |
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|
591 */ |
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|
592 static int play(void* data, int len, int flags) |
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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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593 { |
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|
594 DWORD play_offset; |
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|
595 int space; |
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|
596 |
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|
597 // make sure we have enough space to write data |
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|
598 IDirectSoundBuffer_GetCurrentPosition(hdsbuf,&play_offset,NULL); |
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599 space=buffer_size-(write_offset-play_offset); |
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600 if(space > buffer_size)space -= buffer_size; // write_offset < play_offset |
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|
601 if(space < len) len = space; |
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|
602 |
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|
603 len = (len / ao_data.outburst) * ao_data.outburst; |
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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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604 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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605 } |
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606 |
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607 /** |
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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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608 \brief get the delay between the first and last sample in the buffer |
70d8f1975fc8
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609 \return delay in seconds |
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610 */ |
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611 static float get_delay() |
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612 { |
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613 DWORD play_offset; |
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614 int space; |
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615 IDirectSoundBuffer_GetCurrentPosition(hdsbuf,&play_offset,NULL); |
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616 space=play_offset-write_offset; |
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617 if(space <= 0)space += buffer_size; |
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618 return (float)(buffer_size - space) / (float)ao_data.bps; |
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619 } |