Mercurial > mplayer.hg
annotate libao2/ao_dsound.c @ 28992:947ef23ba798
Test if create_vdp_decoder() might succeed by calling it from config()
with a small value for max_reference_frames.
This does not make automatic recovery by using software decoder possible,
but lets MPlayer fail more graciously on - actually existing - buggy
hardware that does not support certain H264 widths when using
hardware accelerated decoding (784, 864, 944, 1024, 1808, 1888 pixels on
NVIDIA G98) and if the user tries to hardware-decode more samples at
the same time than supported.
Might break playback of H264 Intra-Only samples on hardware with very
little video memory.
author | cehoyos |
---|---|
date | Sat, 21 Mar 2009 20:11:05 +0000 |
parents | 9a5b8c2ed6de |
children | 0f1b5b68af32 |
rev | line source |
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1 /* |
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2 * Windows DirectSound interface |
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3 * |
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4 * Copyright (c) 2004 Gabor Szecsi <deje@miki.hu> |
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5 * |
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6 * This file is part of MPlayer. |
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7 * |
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8 * MPlayer is free software; you can redistribute it and/or modify |
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9 * it under the terms of the GNU General Public License as published by |
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10 * the Free Software Foundation; either version 2 of the License, or |
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11 * (at your option) any later version. |
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12 * |
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13 * MPlayer is distributed in the hope that it will be useful, |
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14 * but WITHOUT ANY WARRANTY; without even the implied warranty of |
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15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the |
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16 * GNU General Public License for more details. |
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17 * |
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18 * You should have received a copy of the GNU General Public License along |
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19 * with MPlayer; if not, write to the Free Software Foundation, Inc., |
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20 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. |
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21 */ |
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22 |
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23 /** |
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24 \todo verify/extend multichannel support |
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25 */ |
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26 |
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27 |
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28 #include <stdio.h> |
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29 #include <stdlib.h> |
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30 #include <windows.h> |
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31 #define DIRECTSOUND_VERSION 0x0600 |
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32 #include <dsound.h> |
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33 #include <math.h> |
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34 |
14479 | 35 #include "config.h" |
14245 | 36 #include "libaf/af_format.h" |
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37 #include "audio_out.h" |
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38 #include "audio_out_internal.h" |
14123 | 39 #include "mp_msg.h" |
40 #include "libvo/fastmemcpy.h" | |
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41 #include "osdep/timer.h" |
14398 | 42 #include "subopt-helper.h" |
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43 |
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44 |
28823 | 45 static const ao_info_t info = |
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46 { |
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47 "Windows DirectSound audio output", |
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48 "dsound", |
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49 "Gabor Szecsi <deje@miki.hu>", |
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50 "" |
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51 }; |
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52 |
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53 LIBAO_EXTERN(dsound) |
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54 |
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55 /** |
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56 \todo use the definitions from the win32 api headers when they define these |
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57 */ |
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58 #if 1 |
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59 #define WAVE_FORMAT_IEEE_FLOAT 0x0003 |
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60 #define WAVE_FORMAT_DOLBY_AC3_SPDIF 0x0092 |
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61 #define WAVE_FORMAT_EXTENSIBLE 0xFFFE |
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62 |
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63 static const GUID KSDATAFORMAT_SUBTYPE_PCM = {0x1,0x0000,0x0010, {0x80,0x00,0x00,0xaa,0x00,0x38,0x9b,0x71}}; |
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64 |
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65 #define SPEAKER_FRONT_LEFT 0x1 |
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66 #define SPEAKER_FRONT_RIGHT 0x2 |
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67 #define SPEAKER_FRONT_CENTER 0x4 |
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68 #define SPEAKER_LOW_FREQUENCY 0x8 |
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69 #define SPEAKER_BACK_LEFT 0x10 |
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70 #define SPEAKER_BACK_RIGHT 0x20 |
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71 #define SPEAKER_FRONT_LEFT_OF_CENTER 0x40 |
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72 #define SPEAKER_FRONT_RIGHT_OF_CENTER 0x80 |
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73 #define SPEAKER_BACK_CENTER 0x100 |
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74 #define SPEAKER_SIDE_LEFT 0x200 |
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75 #define SPEAKER_SIDE_RIGHT 0x400 |
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76 #define SPEAKER_TOP_CENTER 0x800 |
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77 #define SPEAKER_TOP_FRONT_LEFT 0x1000 |
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78 #define SPEAKER_TOP_FRONT_CENTER 0x2000 |
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79 #define SPEAKER_TOP_FRONT_RIGHT 0x4000 |
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80 #define SPEAKER_TOP_BACK_LEFT 0x8000 |
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81 #define SPEAKER_TOP_BACK_CENTER 0x10000 |
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82 #define SPEAKER_TOP_BACK_RIGHT 0x20000 |
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83 #define SPEAKER_RESERVED 0x80000000 |
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84 |
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85 #define DSSPEAKER_HEADPHONE 0x00000001 |
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86 #define DSSPEAKER_MONO 0x00000002 |
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87 #define DSSPEAKER_QUAD 0x00000003 |
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88 #define DSSPEAKER_STEREO 0x00000004 |
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89 #define DSSPEAKER_SURROUND 0x00000005 |
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90 #define DSSPEAKER_5POINT1 0x00000006 |
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91 |
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92 #ifndef _WAVEFORMATEXTENSIBLE_ |
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93 typedef struct { |
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94 WAVEFORMATEX Format; |
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95 union { |
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96 WORD wValidBitsPerSample; /* bits of precision */ |
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97 WORD wSamplesPerBlock; /* valid if wBitsPerSample==0 */ |
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98 WORD wReserved; /* If neither applies, set to zero. */ |
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99 } Samples; |
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100 DWORD dwChannelMask; /* which channels are */ |
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101 /* present in stream */ |
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102 GUID SubFormat; |
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103 } WAVEFORMATEXTENSIBLE, *PWAVEFORMATEXTENSIBLE; |
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104 #endif |
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105 |
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106 #endif |
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107 |
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108 static const int channel_mask[] = { |
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109 SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_LOW_FREQUENCY, |
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110 SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT, |
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111 SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | SPEAKER_LOW_FREQUENCY, |
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112 SPEAKER_FRONT_LEFT | SPEAKER_FRONT_CENTER | SPEAKER_FRONT_RIGHT | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | SPEAKER_LOW_FREQUENCY |
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113 }; |
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114 |
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115 static HINSTANCE hdsound_dll = NULL; ///handle to the dll |
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116 static LPDIRECTSOUND hds = NULL; ///direct sound object |
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117 static LPDIRECTSOUNDBUFFER hdspribuf = NULL; ///primary direct sound buffer |
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118 static LPDIRECTSOUNDBUFFER hdsbuf = NULL; ///secondary direct sound buffer (stream buffer) |
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119 static int buffer_size = 0; ///size in bytes of the direct sound buffer |
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120 static int write_offset = 0; ///offset of the write cursor in the direct sound buffer |
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121 static int min_free_space = 0; ///if the free space is below this value get_space() will return 0 |
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122 ///there will always be at least this amout of free space to prevent |
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123 ///get_space() from returning wrong values when buffer is 100% full. |
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124 ///will be replaced with nBlockAlign in init() |
14398 | 125 static int device_num = 0; ///wanted device number |
126 static GUID device; ///guid of the device | |
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127 |
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128 /***************************************************************************************/ |
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129 |
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130 /** |
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131 \brief output error message |
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132 \param err error code |
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133 \return string with the error message |
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134 */ |
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135 static char * dserr2str(int err) |
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136 { |
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137 switch (err) { |
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138 case DS_OK: return "DS_OK"; |
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139 case DS_NO_VIRTUALIZATION: return "DS_NO_VIRTUALIZATION"; |
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140 case DSERR_ALLOCATED: return "DS_NO_VIRTUALIZATION"; |
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141 case DSERR_CONTROLUNAVAIL: return "DSERR_CONTROLUNAVAIL"; |
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142 case DSERR_INVALIDPARAM: return "DSERR_INVALIDPARAM"; |
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143 case DSERR_INVALIDCALL: return "DSERR_INVALIDCALL"; |
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144 case DSERR_GENERIC: return "DSERR_GENERIC"; |
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145 case DSERR_PRIOLEVELNEEDED: return "DSERR_PRIOLEVELNEEDED"; |
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146 case DSERR_OUTOFMEMORY: return "DSERR_OUTOFMEMORY"; |
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147 case DSERR_BADFORMAT: return "DSERR_BADFORMAT"; |
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148 case DSERR_UNSUPPORTED: return "DSERR_UNSUPPORTED"; |
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149 case DSERR_NODRIVER: return "DSERR_NODRIVER"; |
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150 case DSERR_ALREADYINITIALIZED: return "DSERR_ALREADYINITIALIZED"; |
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151 case DSERR_NOAGGREGATION: return "DSERR_NOAGGREGATION"; |
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152 case DSERR_BUFFERLOST: return "DSERR_BUFFERLOST"; |
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153 case DSERR_OTHERAPPHASPRIO: return "DSERR_OTHERAPPHASPRIO"; |
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154 case DSERR_UNINITIALIZED: return "DSERR_UNINITIALIZED"; |
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155 case DSERR_NOINTERFACE: return "DSERR_NOINTERFACE"; |
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156 case DSERR_ACCESSDENIED: return "DSERR_ACCESSDENIED"; |
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157 default: return "unknown"; |
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158 } |
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159 } |
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160 |
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161 /** |
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162 \brief uninitialize direct sound |
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163 */ |
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164 static void UninitDirectSound(void) |
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165 { |
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166 // finally release the DirectSound object |
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167 if (hds) { |
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168 IDirectSound_Release(hds); |
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169 hds = NULL; |
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170 } |
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171 // free DSOUND.DLL |
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172 if (hdsound_dll) { |
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173 FreeLibrary(hdsound_dll); |
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174 hdsound_dll = NULL; |
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175 } |
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176 mp_msg(MSGT_AO, MSGL_V, "ao_dsound: DirectSound uninitialized\n"); |
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177 } |
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178 |
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179 /** |
14398 | 180 \brief print the commandline help |
181 */ | |
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182 static void print_help(void) |
14398 | 183 { |
184 mp_msg(MSGT_AO, MSGL_FATAL, | |
185 "\n-ao dsound commandline help:\n" | |
186 "Example: mplayer -ao dsound:device=1\n" | |
187 " sets 1st device\n" | |
188 "\nOptions:\n" | |
189 " device=<device-number>\n" | |
190 " Sets device number, use -v to get a list\n"); | |
191 } | |
192 | |
193 | |
194 /** | |
195 \brief enumerate direct sound devices | |
196 \return TRUE to continue with the enumeration | |
197 */ | |
198 static BOOL CALLBACK DirectSoundEnum(LPGUID guid,LPCSTR desc,LPCSTR module,LPVOID context) | |
199 { | |
200 int* device_index=context; | |
201 mp_msg(MSGT_AO, MSGL_V,"%i %s ",*device_index,desc); | |
202 if(device_num==*device_index){ | |
203 mp_msg(MSGT_AO, MSGL_V,"<--"); | |
204 if(guid){ | |
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205 memcpy(&device,guid,sizeof(GUID)); |
14398 | 206 } |
207 } | |
208 mp_msg(MSGT_AO, MSGL_V,"\n"); | |
209 (*device_index)++; | |
210 return TRUE; | |
211 } | |
212 | |
213 | |
214 /** | |
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215 \brief initilize direct sound |
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216 \return 0 if error, 1 if ok |
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217 */ |
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218 static int InitDirectSound(void) |
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219 { |
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220 DSCAPS dscaps; |
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221 |
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222 // initialize directsound |
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223 HRESULT (WINAPI *OurDirectSoundCreate)(LPGUID, LPDIRECTSOUND *, LPUNKNOWN); |
14398 | 224 HRESULT (WINAPI *OurDirectSoundEnumerate)(LPDSENUMCALLBACKA, LPVOID); |
225 int device_index=0; | |
226 opt_t subopts[] = { | |
227 {"device", OPT_ARG_INT, &device_num,NULL}, | |
228 {NULL} | |
229 }; | |
230 if (subopt_parse(ao_subdevice, subopts) != 0) { | |
231 print_help(); | |
232 return 0; | |
233 } | |
234 | |
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235 hdsound_dll = LoadLibrary("DSOUND.DLL"); |
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236 if (hdsound_dll == NULL) { |
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237 mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: cannot load DSOUND.DLL\n"); |
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238 return 0; |
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239 } |
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240 OurDirectSoundCreate = (void*)GetProcAddress(hdsound_dll, "DirectSoundCreate"); |
14398 | 241 OurDirectSoundEnumerate = (void*)GetProcAddress(hdsound_dll, "DirectSoundEnumerateA"); |
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242 |
14398 | 243 if (OurDirectSoundCreate == NULL || OurDirectSoundEnumerate == NULL) { |
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244 mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: GetProcAddress FAILED\n"); |
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245 FreeLibrary(hdsound_dll); |
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246 return 0; |
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247 } |
14398 | 248 |
249 // Enumerate all directsound devices | |
250 mp_msg(MSGT_AO, MSGL_V,"ao_dsound: Output Devices:\n"); | |
251 OurDirectSoundEnumerate(DirectSoundEnum,&device_index); | |
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252 |
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253 // Create the direct sound object |
14398 | 254 if FAILED(OurDirectSoundCreate((device_num)?&device:NULL, &hds, NULL )) { |
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255 mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: cannot create a DirectSound device\n"); |
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256 FreeLibrary(hdsound_dll); |
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257 return 0; |
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258 } |
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259 |
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260 /* Set DirectSound Cooperative level, ie what control we want over Windows |
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261 * sound device. In our case, DSSCL_EXCLUSIVE means that we can modify the |
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262 * settings of the primary buffer, but also that only the sound of our |
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263 * application will be hearable when it will have the focus. |
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264 * !!! (this is not really working as intended yet because to set the |
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265 * cooperative level you need the window handle of your application, and |
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266 * I don't know of any easy way to get it. Especially since we might play |
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267 * sound without any video, and so what window handle should we use ??? |
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268 * The hack for now is to use the Desktop window handle - it seems to be |
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269 * working */ |
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270 if (IDirectSound_SetCooperativeLevel(hds, GetDesktopWindow(), DSSCL_EXCLUSIVE)) { |
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271 mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: cannot set direct sound cooperative level\n"); |
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272 IDirectSound_Release(hds); |
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273 FreeLibrary(hdsound_dll); |
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274 return 0; |
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275 } |
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276 mp_msg(MSGT_AO, MSGL_V, "ao_dsound: DirectSound initialized\n"); |
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277 |
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278 memset(&dscaps, 0, sizeof(DSCAPS)); |
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279 dscaps.dwSize = sizeof(DSCAPS); |
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280 if (DS_OK == IDirectSound_GetCaps(hds, &dscaps)) { |
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281 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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282 } else { |
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283 mp_msg(MSGT_AO, MSGL_V, "ao_dsound: cannot get device capabilities\n"); |
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284 } |
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285 |
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286 return 1; |
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287 } |
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288 |
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289 /** |
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290 \brief destroy the direct sound buffer |
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291 */ |
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292 static void DestroyBuffer(void) |
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293 { |
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294 if (hdsbuf) { |
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295 IDirectSoundBuffer_Release(hdsbuf); |
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296 hdsbuf = NULL; |
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297 } |
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298 if (hdspribuf) { |
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299 IDirectSoundBuffer_Release(hdspribuf); |
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300 hdspribuf = NULL; |
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301 } |
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302 } |
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303 |
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304 /** |
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305 \brief fill sound buffer |
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306 \param data pointer to the sound data to copy |
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307 \param len length of the data to copy in bytes |
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308 \return number of copyed bytes |
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309 */ |
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310 static int write_buffer(unsigned char *data, int len) |
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311 { |
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312 HRESULT res; |
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313 LPVOID lpvPtr1; |
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314 DWORD dwBytes1; |
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315 LPVOID lpvPtr2; |
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316 DWORD dwBytes2; |
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317 |
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318 // Lock the buffer |
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319 res = IDirectSoundBuffer_Lock(hdsbuf,write_offset, len, &lpvPtr1, &dwBytes1, &lpvPtr2, &dwBytes2, 0); |
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320 // If the buffer was lost, restore and retry lock. |
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321 if (DSERR_BUFFERLOST == res) |
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322 { |
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323 IDirectSoundBuffer_Restore(hdsbuf); |
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324 res = IDirectSoundBuffer_Lock(hdsbuf,write_offset, len, &lpvPtr1, &dwBytes1, &lpvPtr2, &dwBytes2, 0); |
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325 } |
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326 |
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327 |
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328 if (SUCCEEDED(res)) |
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329 { |
14245 | 330 if( (ao_data.channels == 6) && (ao_data.format!=AF_FORMAT_AC3) ) { |
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331 // reorder channels while writing to pointers. |
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332 // it's this easy because buffer size and len are always |
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333 // aligned to multiples of channels*bytespersample |
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334 // there's probably some room for speed improvements here |
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335 const int chantable[6] = {0, 1, 4, 5, 2, 3}; // reorder "matrix" |
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336 int i, j; |
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337 int numsamp,sampsize; |
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338 |
14249 | 339 sampsize = af_fmt2bits(ao_data.format)>>3; // bytes per sample |
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340 numsamp = dwBytes1 / (ao_data.channels * sampsize); // number of samples for each channel in this buffer |
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341 |
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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(lpvPtr1+(i*ao_data.channels*sampsize)+(chantable[j]*sampsize),data+(i*ao_data.channels*sampsize)+(j*sampsize),sampsize); |
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344 } |
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345 |
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346 if (NULL != lpvPtr2 ) |
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347 { |
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348 numsamp = dwBytes2 / (ao_data.channels * sampsize); |
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349 for( i = 0; i < numsamp; i++ ) for( j = 0; j < ao_data.channels; j++ ) { |
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350 memcpy(lpvPtr2+(i*ao_data.channels*sampsize)+(chantable[j]*sampsize),data+dwBytes1+(i*ao_data.channels*sampsize)+(j*sampsize),sampsize); |
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351 } |
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352 } |
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353 |
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354 write_offset+=dwBytes1+dwBytes2; |
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355 if(write_offset>=buffer_size)write_offset=dwBytes2; |
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356 } else { |
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357 // Write to pointers without reordering. |
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358 fast_memcpy(lpvPtr1,data,dwBytes1); |
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359 if (NULL != lpvPtr2 )fast_memcpy(lpvPtr2,data+dwBytes1,dwBytes2); |
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360 write_offset+=dwBytes1+dwBytes2; |
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361 if(write_offset>=buffer_size)write_offset=dwBytes2; |
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362 } |
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363 |
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364 // Release the data back to DirectSound. |
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365 res = IDirectSoundBuffer_Unlock(hdsbuf,lpvPtr1,dwBytes1,lpvPtr2,dwBytes2); |
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366 if (SUCCEEDED(res)) |
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367 { |
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368 // Success. |
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369 DWORD status; |
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370 IDirectSoundBuffer_GetStatus(hdsbuf, &status); |
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371 if (!(status & DSBSTATUS_PLAYING)){ |
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372 res = IDirectSoundBuffer_Play(hdsbuf, 0, 0, DSBPLAY_LOOPING); |
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373 } |
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374 return dwBytes1+dwBytes2; |
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375 } |
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376 } |
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377 // Lock, Unlock, or Restore failed. |
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378 return 0; |
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379 } |
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380 |
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381 /***************************************************************************************/ |
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382 |
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383 /** |
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384 \brief handle control commands |
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385 \param cmd command |
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386 \param arg argument |
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387 \return CONTROL_OK or -1 in case the command can't be handled |
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388 */ |
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389 static int control(int cmd, void *arg) |
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390 { |
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391 DWORD volume; |
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392 switch (cmd) { |
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393 case AOCONTROL_GET_VOLUME: { |
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394 ao_control_vol_t* vol = (ao_control_vol_t*)arg; |
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395 IDirectSoundBuffer_GetVolume(hdsbuf, &volume); |
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396 vol->left = vol->right = pow(10.0, (float)(volume+10000) / 5000.0); |
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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 case AOCONTROL_SET_VOLUME: { |
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401 ao_control_vol_t* vol = (ao_control_vol_t*)arg; |
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402 volume = (DWORD)(log10(vol->right) * 5000.0) - 10000; |
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403 IDirectSoundBuffer_SetVolume(hdsbuf, volume); |
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404 //printf("ao_dsound: volume: %f\n",vol->left); |
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405 return CONTROL_OK; |
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406 } |
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407 } |
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408 return -1; |
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409 } |
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410 |
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411 /** |
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412 \brief setup sound device |
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413 \param rate samplerate |
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414 \param channels number of channels |
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415 \param format format |
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416 \param flags unused |
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417 \return 1=success 0=fail |
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418 */ |
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419 static int init(int rate, int channels, int format, int flags) |
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420 { |
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421 int res; |
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422 if (!InitDirectSound()) return 0; |
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423 |
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424 // ok, now create the buffers |
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425 WAVEFORMATEXTENSIBLE wformat; |
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426 DSBUFFERDESC dsbpridesc; |
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427 DSBUFFERDESC dsbdesc; |
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428 |
14135 | 429 //check if the format is supported in general |
430 switch(format){ | |
14245 | 431 case AF_FORMAT_AC3: |
432 case AF_FORMAT_S24_LE: | |
433 case AF_FORMAT_S16_LE: | |
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434 case AF_FORMAT_U8: |
14135 | 435 break; |
436 default: | |
14264 | 437 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 | 438 format=AF_FORMAT_S16_LE; |
14135 | 439 } |
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440 //fill global ao_data |
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441 ao_data.channels = channels; |
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442 ao_data.samplerate = rate; |
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443 ao_data.format = format; |
14249 | 444 ao_data.bps = channels * rate * (af_fmt2bits(format)>>3); |
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445 if(ao_data.buffersize==-1) ao_data.buffersize = ao_data.bps; // space for 1 sec |
14264 | 446 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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447 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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448 |
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449 //fill waveformatex |
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450 ZeroMemory(&wformat, sizeof(WAVEFORMATEXTENSIBLE)); |
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451 wformat.Format.cbSize = (channels > 2) ? sizeof(WAVEFORMATEXTENSIBLE)-sizeof(WAVEFORMATEX) : 0; |
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452 wformat.Format.nChannels = channels; |
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453 wformat.Format.nSamplesPerSec = rate; |
14245 | 454 if (format == AF_FORMAT_AC3) { |
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455 wformat.Format.wFormatTag = WAVE_FORMAT_DOLBY_AC3_SPDIF; |
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456 wformat.Format.wBitsPerSample = 16; |
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457 wformat.Format.nBlockAlign = 4; |
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458 } else { |
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459 wformat.Format.wFormatTag = (channels > 2) ? WAVE_FORMAT_EXTENSIBLE : WAVE_FORMAT_PCM; |
14249 | 460 wformat.Format.wBitsPerSample = af_fmt2bits(format); |
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461 wformat.Format.nBlockAlign = wformat.Format.nChannels * (wformat.Format.wBitsPerSample >> 3); |
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462 } |
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463 |
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464 // fill in primary sound buffer descriptor |
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465 memset(&dsbpridesc, 0, sizeof(DSBUFFERDESC)); |
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466 dsbpridesc.dwSize = sizeof(DSBUFFERDESC); |
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467 dsbpridesc.dwFlags = DSBCAPS_PRIMARYBUFFER; |
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468 dsbpridesc.dwBufferBytes = 0; |
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469 dsbpridesc.lpwfxFormat = NULL; |
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470 |
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471 |
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472 // fill in the secondary sound buffer (=stream buffer) descriptor |
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473 memset(&dsbdesc, 0, sizeof(DSBUFFERDESC)); |
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474 dsbdesc.dwSize = sizeof(DSBUFFERDESC); |
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475 dsbdesc.dwFlags = DSBCAPS_GETCURRENTPOSITION2 /** Better position accuracy */ |
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476 | DSBCAPS_GLOBALFOCUS /** Allows background playing */ |
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477 | DSBCAPS_CTRLVOLUME; /** volume control enabled */ |
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478 |
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479 if (channels > 2) { |
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480 wformat.dwChannelMask = channel_mask[channels - 3]; |
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481 wformat.SubFormat = KSDATAFORMAT_SUBTYPE_PCM; |
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482 wformat.Samples.wValidBitsPerSample = wformat.Format.wBitsPerSample; |
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483 // Needed for 5.1 on emu101k - shit soundblaster |
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484 dsbdesc.dwFlags |= DSBCAPS_LOCHARDWARE; |
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485 } |
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486 wformat.Format.nAvgBytesPerSec = wformat.Format.nSamplesPerSec * wformat.Format.nBlockAlign; |
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487 |
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488 dsbdesc.dwBufferBytes = ao_data.buffersize; |
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489 dsbdesc.lpwfxFormat = (WAVEFORMATEX *)&wformat; |
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490 buffer_size = dsbdesc.dwBufferBytes; |
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491 write_offset = 0; |
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492 min_free_space = wformat.Format.nBlockAlign; |
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493 ao_data.outburst = wformat.Format.nBlockAlign * 512; |
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494 |
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495 // create primary buffer and set its format |
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496 |
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497 res = IDirectSound_CreateSoundBuffer( hds, &dsbpridesc, &hdspribuf, NULL ); |
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498 if ( res != DS_OK ) { |
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499 UninitDirectSound(); |
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500 mp_msg(MSGT_AO, MSGL_ERR,"ao_dsound: cannot create primary buffer (%s)\n", dserr2str(res)); |
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501 return 0; |
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502 } |
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503 res = IDirectSoundBuffer_SetFormat( hdspribuf, (WAVEFORMATEX *)&wformat ); |
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504 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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505 |
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506 mp_msg(MSGT_AO, MSGL_V, "ao_dsound: primary buffer created\n"); |
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507 |
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508 // now create the stream buffer |
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509 |
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510 res = IDirectSound_CreateSoundBuffer(hds, &dsbdesc, &hdsbuf, NULL); |
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511 if (res != DS_OK) { |
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512 if (dsbdesc.dwFlags & DSBCAPS_LOCHARDWARE) { |
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513 // Try without DSBCAPS_LOCHARDWARE |
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514 dsbdesc.dwFlags &= ~DSBCAPS_LOCHARDWARE; |
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515 res = IDirectSound_CreateSoundBuffer(hds, &dsbdesc, &hdsbuf, NULL); |
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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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516 } |
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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 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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518 UninitDirectSound(); |
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519 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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520 return 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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521 } |
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|
522 } |
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523 mp_msg(MSGT_AO, MSGL_V, "ao_dsound: secondary (stream)buffer created\n"); |
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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 return 1; |
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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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directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
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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 |
70d8f1975fc8
directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
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|
528 |
70d8f1975fc8
directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
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529 /** |
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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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|
530 \brief stop playing and empty buffers (for seeking/pause) |
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531 */ |
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532 static void reset(void) |
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533 { |
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534 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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535 // 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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536 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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537 write_offset=0; |
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538 } |
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539 |
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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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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 stop playing, keep buffers (for pause) |
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542 */ |
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543 static void audio_pause(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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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 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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546 } |
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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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|
547 |
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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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|
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 resume playing, after audio_pause() |
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550 */ |
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|
551 static void audio_resume(void) |
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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 IDirectSoundBuffer_Play(hdsbuf, 0, 0, DSBPLAY_LOOPING); |
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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 } |
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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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555 |
70d8f1975fc8
directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
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|
556 /** |
70d8f1975fc8
directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
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|
557 \brief close audio device |
14555
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558 \param immed stop playback immediately |
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559 */ |
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560 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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561 { |
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|
562 if(immed)reset(); |
6fb9eca97d41
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563 else{ |
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564 DWORD status; |
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565 IDirectSoundBuffer_Play(hdsbuf, 0, 0, 0); |
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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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566 while(!IDirectSoundBuffer_GetStatus(hdsbuf,&status) && (status&DSBSTATUS_PLAYING)) |
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567 usec_sleep(20000); |
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568 } |
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569 DestroyBuffer(); |
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570 UninitDirectSound(); |
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571 } |
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572 |
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573 /** |
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574 \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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575 \return free space in bytes, has to return 0 if the buffer is almost full |
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576 */ |
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577 static int get_space(void) |
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578 { |
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579 int space; |
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580 DWORD play_offset; |
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581 IDirectSoundBuffer_GetCurrentPosition(hdsbuf,&play_offset,NULL); |
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582 space=buffer_size-(write_offset-play_offset); |
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|
583 // | | <-- const --> | | | |
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584 // buffer start play_cursor write_cursor write_offset buffer end |
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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 // play_cursor is the actual postion of the play cursor |
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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 // write_cursor is the position after which it is assumed to be save to write data |
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587 // write_offset is the postion where we actually write the data to |
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588 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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|
589 if(space < min_free_space)return 0; |
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diff
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|
590 return space-min_free_space; |
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|
591 } |
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|
592 |
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faust3
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|
593 /** |
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faust3
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594 \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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|
595 \param data pointer to the data to play |
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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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596 \param len size in bytes of the data buffer, gets rounded down to outburst*n |
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597 \param flags currently unused |
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598 \return number of played bytes |
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|
599 */ |
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600 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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|
601 { |
13674
b7322244e53c
channel reorder patch by Florian Dietrich <flodt8 at yahoo.de>
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|
602 DWORD play_offset; |
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channel reorder patch by Florian Dietrich <flodt8 at yahoo.de>
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|
603 int space; |
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|
604 |
b7322244e53c
channel reorder patch by Florian Dietrich <flodt8 at yahoo.de>
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|
605 // make sure we have enough space to write data |
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13640
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|
606 IDirectSoundBuffer_GetCurrentPosition(hdsbuf,&play_offset,NULL); |
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faust3
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607 space=buffer_size-(write_offset-play_offset); |
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faust3
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608 if(space > buffer_size)space -= buffer_size; // write_offset < play_offset |
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|
609 if(space < len) len = space; |
b7322244e53c
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|
610 |
20250
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Implement AOPLAY_FINAL_CHUNK support for dshow and win32 out.
reimar
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19614
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|
611 if (!(flags & AOPLAY_FINAL_CHUNK)) |
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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 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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613 return write_buffer(data, len); |
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|
614 } |
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|
615 |
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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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directsound audio output plugin, patch by Gabor Szecsi <deje at miki.hu> some minor modifications by me
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|
617 \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
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618 \return delay in seconds |
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619 */ |
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|
620 static float get_delay(void) |
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621 { |
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|
622 DWORD play_offset; |
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|
623 int space; |
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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 IDirectSoundBuffer_GetCurrentPosition(hdsbuf,&play_offset,NULL); |
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625 space=play_offset-write_offset; |
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626 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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627 return (float)(buffer_size - space) / (float)ao_data.bps; |
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628 } |