Mercurial > mplayer.hg
annotate DOCS/sound.html @ 9166:c8771ae299fb
use MASTER instead of PCM
author | gabucino |
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date | Wed, 29 Jan 2003 22:20:36 +0000 |
parents | 68efb63884b2 |
children | eeca9a1043cd |
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1 <!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> |
1612 | 2 <HTML> |
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3 |
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4 <HEAD> |
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5 <TITLE>Sound - MPlayer - The Movie Player for Linux</TITLE> |
6391 | 6 <LINK REL="stylesheet" TYPE="text/css" HREF="default.css"> |
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7 <META HTTP-EQUIV="Content-Type" CONTENT="text/html; charset=iso-8859-1"> |
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8 </HEAD> |
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9 |
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10 <BODY> |
1704 | 11 |
1612 | 12 |
7047 | 13 <H3><A NAME="audio">2.3.2 Audio output devices</A></H3> |
1612 | 14 |
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15 <H4><A NAME="sync">2.3.2.1 Audio/Video synchronisation</A></H4> |
8177 | 16 |
7814 | 17 <P>MPlayer's audio interface is called <I>libao2</I>. It currently |
6974 | 18 contains these drivers:</P> |
1892 | 19 |
8751 | 20 <DL> |
21 <DT>oss</DT> | |
22 <DD>OSS (ioctl) driver (supports hardware AC3 passthrough)</DD> | |
23 | |
24 <DT>sdl</DT> | |
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25 <DD>SDL driver (supports sound daemons like <B>ESD</B> and <B>ARTS</B>)</DD> |
8751 | 26 |
27 <DT>nas</DT> | |
28 <DD>NAS (Network Audio System) driver</DD> | |
29 | |
30 <DT>alsa5</DT> | |
31 <DD>native ALSA 0.5 driver</DD> | |
32 | |
33 <DT>alsa9</DT> | |
34 <DD>native ALSA 0.9 driver (supports hardware AC3 passthrough)</DD> | |
35 | |
36 <DT>sun</DT> | |
37 <DD>SUN audio driver (<CODE>/dev/audio</CODE>) for BSD and Solaris8 users</DD> | |
38 | |
39 <DT>arts</DT> | |
40 <DD>native ARTS driver (mostly for KDE users)</DD> | |
41 | |
42 <DT>esd</DT> | |
43 <DD>native ESD driver (mostly for GNOME users)</DD> | |
44 </DL> | |
1612 | 45 |
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46 <P>Linux sound card drivers have compatibility problems. This is because MPlayer |
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47 relies on an in-built feature of <EM>properly</EM> coded sound drivers that |
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48 enable them to maintain correct audio/video sync. Regrettably, some driver |
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49 authors don't take the care to code this feature since it is not needed for |
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50 playing MP3s or sound effects. </P> |
8177 | 51 |
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52 <P>Other media players like <A HREF="http://avifile.sourceforge.net">aviplay</A> |
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53 or <A HREF="http://xine.sourceforge.net">xine</A> possibly work |
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54 out-of-the-box with these drivers because they use "simple" methods with |
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55 internal timing. Measuring showed that their methods are not as efficient |
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56 as MPlayer's. </P> |
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57 |
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58 <P>Using MPlayer with a properly written audio driver will never result |
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59 in A/V desyncs related to the audio, except only with very badly created |
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60 files (check the man page for workarounds).</P> |
8177 | 61 |
62 <P>If you happen to have a bad audio driver, try the <CODE>-autosync</CODE> | |
63 option, it should sort out your problems. See the man page for detailed | |
64 information.</P> | |
65 | |
66 <P>Some notes:</P> | |
5765 | 67 |
68 <UL> | |
6960 | 69 <LI>If you have an OSS driver, first try <CODE>-ao oss</CODE> (this is the |
70 default). If you experience glitches, halts or anything out of the | |
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71 ordinary, try <CODE>-ao sdl</CODE> (NOTE: You need to have SDL libraries |
6960 | 72 and header files installed). The SDL audio driver helps in a lot of cases |
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73 and also supports ESD (GNOME) and ARTS (KDE).</LI> |
6960 | 74 <LI>If you have ALSA version 0.5, then you almost always have to use |
75 <CODE>-ao alsa5</CODE> , since ALSA 0.5 has buggy OSS emulation code, and | |
76 will <B>crash MPlayer</B> with a message like this:<BR> | |
77 <CODE>DEMUXER: Too many (945 in 8390980 bytes) video packets in the buffer!</CODE></LI> | |
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78 <LI>On Solaris, use the SUN audio driver with the <CODE>-ao sun</CODE> option, |
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79 otherwise neither video nor audio will work.</LI> |
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80 <LI>If the sound clicks when playing from CD-ROM, turn on IRQ unmasking, e.g. |
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81 <CODE>hdparm -u1 /dev/cdrom</CODE> (<CODE>man hdparm</CODE>). This is |
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82 generally beneficial and described in more detail in the |
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83 <A HREF="cd-dvd.html#drives">CD-ROM section</A>.</LI> |
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84 </UL> |
1892 | 85 |
7047 | 86 |
8177 | 87 <H4><A NAME="experiences">2.3.2.2 Sound Card experiences, recommendations</A></H4> |
1612 | 88 |
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89 <P>On Linux, a 2.4.x kernel is highly recommended. Kernel 2.2 is not tested.</P> |
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90 |
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91 <P>Linux sound drivers are primarily provided by the free version of OSS. These |
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92 drivers have been superceded by <A HREF="http://www.alsa-project.org">ALSA</A> |
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93 (Advanced Linux Sound Architecture) in the 2.5 development series. If your |
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94 distribution does not already use ALSA you may wish to try their drivers if |
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95 you experience sound problems. ALSA drivers are generally superior to OSS in |
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96 compatibility, performance and features. But some sound cards are only |
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97 supported by the commercial OSS drivers from |
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98 <A HREF="http://www.opensound.com/">4Front Technologies</A>. They also support |
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99 several non-Linux systems.</P> |
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100 |
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101 <TABLE BORDER="1" WIDTH="100%"> |
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102 |
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103 <TR> |
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104 <TH ROWSPAN="2"><B>SOUND CARD</B></TH> |
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105 <TH COLSPAN="4"><B>DRIVER</B></TH> |
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106 <TH ROWSPAN="2"><B>Max kHz</B></TH> |
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107 </TR> |
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108 |
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109 <TR> |
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110 <TH><B>OSS/Free</B></TH> |
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111 <TH><B>ALSA</B></TH> |
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112 <TH><B>OSS/Pro</B></TH> |
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113 <TH><B>other</B></TH> |
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114 </TR> |
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115 |
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116 <TR> |
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117 <TD><B>VIA onboard (686/A/B, 8233, 8235)</B></TD> |
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118 <TD><A HREF="http://sourceforge.net/project/showfiles.php?group_id=3242&release_id=59602">via82cxxx_audio</A></TD> |
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119 <TD>snd-via82xx</TD> |
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120 <TD> </TD> |
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121 <TD> </TD> |
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122 <TD>4-48 kHz or 48 kHz only, depending on the chipset</TD> |
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123 </TR> |
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124 |
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125 <TR> |
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126 <TD><B>Aureal Vortex 2</B></TD> |
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127 <TD>none</TD> |
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128 <TD>none</TD> |
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129 <TD>OK</TD> |
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130 <TD><A HREF="http://aureal.sourceforge.net">Linux Aureal Drivers</A><BR> |
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131 <A HREF="http://makacs.poliod.hu/~pontscho/aureal/au88xx-1.1.3.tar.bz2">buffer size increased to 32k</A></TD> |
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132 <TD>48</TD> |
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133 </TR> |
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134 |
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135 <TR> |
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136 <TD><B>GUS PnP</B></TD> |
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137 <TD>none</TD> |
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138 <TD>OK</TD> |
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139 <TD>OK</TD> |
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140 <TD> </TD> |
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141 <TD>48</TD> |
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142 </TR> |
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143 |
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144 <TR> |
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145 <TD><B>SB Live!</B></TD> |
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146 <TD>Analog OK, SP/DIF not working</TD> |
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147 <TD>Both OK</TD> |
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148 <TD> </TD> |
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149 <TD> </TD> |
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150 <TD>192</TD> |
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151 </TR> |
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152 |
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153 <TR> |
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154 <TD><B>SB AWE 64</B></TD> |
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155 <TD>max 44kHz</TD> |
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156 <TD>48kHz sounds bad</TD> |
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157 <TD> </TD> |
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158 <TD> </TD> |
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159 <TD>48</TD> |
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160 </TR> |
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161 |
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162 <TR> |
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163 <TD><B>Gravis UltraSound ACE</B></TD> |
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164 <TD>not OK</TD> |
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165 <TD>OK</TD> |
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166 <TD> </TD> |
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167 <TD> </TD> |
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168 <TD>44</TD> |
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169 </TR> |
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170 |
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171 <TR> |
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172 <TD><B>Gravis UltraSound MAX</B></TD> |
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173 <TD>OK</TD> |
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174 <TD>OK (?)</TD> |
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175 <TD> </TD> |
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176 <TD> </TD> |
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177 <TD>48</TD> |
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178 </TR> |
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179 |
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180 <TR> |
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181 <TD><B>ESS 688</B></TD> |
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182 <TD>OK</TD> |
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183 <TD>OK (?)</TD> |
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184 <TD> </TD> |
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185 <TD> </TD> |
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186 <TD>48</TD> |
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187 </TR> |
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188 |
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189 <TR> |
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190 <TD><B>C-Media cards (which ones?)</B></TD> |
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191 <TD>not OK (hissing) (?)</TD> |
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192 <TD>OK (?)</TD> |
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193 <TD> </TD> |
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194 <TD> </TD> |
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195 <TD>?</TD> |
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196 </TR> |
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197 |
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198 <TR> |
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199 <TD><B>Yamaha cards (*ymf*)</B></TD> |
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200 <TD>not OK (?) (maybe <CODE>-ao sdl</CODE>)</TD> |
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201 <TD>OK only with ALSA 0.5 with OSS emulation <B>AND</B> |
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202 <CODE>-ao sdl</CODE> (!) (?)</TD> |
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203 <TD> </TD> |
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204 <TD> </TD> |
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205 <TD>?</TD> |
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206 </TR> |
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207 |
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208 <TR> |
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209 <TD><B>Cards with envy24 chips (like Terratec EWS88MT)</B></TD> |
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210 <TD>?</TD> |
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211 <TD>?</TD> |
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212 <TD>OK</TD> |
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213 <TD> </TD> |
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214 <TD>?</TD> |
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215 </TR> |
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216 |
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217 <TR> |
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218 <TD><B>PC Speaker or DAC</B></TD> |
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219 <TD>OK (Use the SDL driver: <CODE>-ao sdl</CODE>)</TD> |
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220 <TD>none</TD> |
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221 <TD> </TD> |
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222 <TD><A HREF="http://www.geocities.com/stssppnn/pcsp.html">Linux PC speaker OSS driver</a></TD> |
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223 <TD>The driver emulates 44.1, maybe more.</TD> |
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224 </TR> |
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225 |
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226 </TABLE> |
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227 |
7814 | 228 <P>Feedback to this document is welcome. Please tell us how MPlayer |
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229 and your sound card(s) worked together.</P> |
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3671 | 231 |
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232 <H4><A NAME="af">2.3.2.3 Audio filters</A></H4> |
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233 |
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234 <P>The old audio plugins have been superseded by a new audio filter layer. Audio |
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235 filters are used for changing the properties of the audio data before the |
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236 sound reaches the sound card. The activation and deactivation of the filters |
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237 is normally automated but can be overridden. The filters are activated when |
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238 the properties of the audio data differ from those required by the sound card |
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239 and deactivated if unnecessary. The <CODE>-af filter1,filter2,...</CODE> |
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240 option is used to override the automatic activation of filters or to insert |
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241 filters that are not automatically inserted. The filters will be executed as |
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242 they appear in the comma separated list.</P> |
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243 |
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244 <P>Example:<BR> |
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245 <CODE>mplayer -af resample,pan movie.avi </CODE></P> |
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246 |
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247 <P>would run the sound through the resampling filter followed by the pan filter. |
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248 Observe that the list must not contain any spaces, else it will fail.</P> |
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249 |
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250 <P>The filters often have options that change their behavior. These options |
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251 are explained in detail in the sections below. A filter will execute using |
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252 default settings if its options are omitted. Here is an example of how to use |
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253 filters in combination with filter specific options:</P> |
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254 |
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255 <P> <CODE>mplayer -af resample=11025,pan=1:0.5:0.5 -channels 1 |
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256 -srate 11025 media.avi</CODE></P> |
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257 |
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258 <P>would set the output frequency of the resample filter to 11025Hz and downmix |
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259 the audio to 1 channel using the pan filter.</P> |
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260 |
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261 <P>The overall execution of the filter layer is controlled using the |
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262 <CODE>-af-adv</CODE> option. This option has two suboptions:</P> |
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263 |
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264 <DL> |
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265 <DT><CODE>force</CODE><DT> |
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266 <DD>is a Bit field that controls how the filters are inserted and what |
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267 speed/accuracy optimizations they use: |
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268 <DL> |
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269 <DT><CODE>0</CODE></DT> |
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270 <DD>Use automatic insertion of filters and optimize according to CPU |
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271 speed.</DD> |
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272 <DT><CODE>1</CODE></DT> |
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273 <DD>Use automatic insertion of filters and optimize for the highest |
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274 speed.<BR> |
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275 <EM>Warning:</EM> Some features in the audio filters may silently fail, |
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276 and the sound quality may drop.</DD> |
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277 <DT><CODE>2</CODE></DT> |
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278 <DD>Use automatic insertion of filters and optimize for quality.</DD> |
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279 <DT><CODE>3</CODE></DT> |
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280 <DD>Use no automatic insertion of filters and no optimization.<BR> |
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281 <I>Warning:</I> It may be possible to crash MPlayer using this |
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282 setting.</DD> |
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283 <DT><CODE>4</CODE></DT> |
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284 <DD>Use automatic insertion of filters according to 0 above, but use |
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285 floating point processing when possible.</DD> |
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286 <DT><CODE>5</CODE></DT> |
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287 <DD>Use automatic insertion of filters according to 1 above, but use |
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288 floating point processing when possible.</DD> |
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289 <DT><CODE>6</CODE></DT> |
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290 <DD>Use automatic insertion of filters according to 2 above, but use |
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291 floating point processing when possible.</DD> |
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292 <DT><CODE>7</CODE></DT> |
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293 <DD>Use no automatic insertion of filters according to 3 above, and use |
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294 floating point processing when possible.</DD> |
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295 </DL> |
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296 </DD> |
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297 |
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298 <DT><CODE>list</CODE></DT> |
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299 <DD>is an alias for the -af option.</DD> |
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300 </DL> |
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301 |
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302 <P>The filter layer is also affected by the following generic options: |
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303 |
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304 <DL> |
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305 <DT><CODE>-v</CODE></DT> |
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306 <DD>Increases the verbosity level and makes most filters print out extra |
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307 status messages.</DD> |
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308 <DT><CODE>-channels</CODE></DT> |
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309 <DD>This option sets the number of output channels you would like your |
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310 sound card to use. |
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311 It also affects the number of channels that are being decoded from the |
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312 media. If the media contains less channels than requested the channels |
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313 filter (see below) will automatically be inserted. The routing will be the |
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314 default routing for the channels filter.</DD> |
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315 <DT><CODE>-srate</CODE></DT> |
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316 <DD>This option selects the sample rate you would like your sound card to |
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317 use (of course the cards have limits on this). If the sample |
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318 frequency of your sound card is different from that of the current media, |
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319 the resample filter (see below) will be inserted into the audio filter layer |
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320 to compensate for the difference.</DD> |
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321 <DT><CODE>-format</CODE><DT> |
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322 <DD>This option sets the sample format between the audio filter layer and the sound |
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323 card. If the requested sample format of your sound card is different from |
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324 that of the current media, a format filter (see below) will be inserted to |
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325 rectify the difference.</DD> |
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326 </DL> |
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327 |
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328 |
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329 <H5><A NAME="af_resample">2.3.2.3.1 Up/Down-sampling</A></H5> |
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330 |
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332 <CODE>resample</CODE> filter. It can be used if you | |
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333 have a fixed frequency sound card or if you are stuck with an old sound card |
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334 that is only capable of max 44.1kHz. This filter is automatically enabled if |
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335 it is necessary, but it can also be explicitly enabled on the command line. It |
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336 has three options:</P> |
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337 |
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338 <DL> |
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340 <DD>is an integer used for setting the output sample |
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341 frequency in Hz. The valid range for this parameter is 8kHz to 192kHz. If |
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342 the input and output sample frequency are the same or if this parameter is |
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343 omitted the filter is automatically unloaded. A high sample frequency |
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344 normally improves the audio quality, especially when used in combination |
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345 with other filters.</DD> |
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346 |
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347 <DT><CODE>sloppy</CODE></DT> |
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348 <DD>is an optional binary parameter that allows the output frequency to differ |
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349 slightly from the frequency given by <CODE>srate</CODE>. This option can be |
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350 used if the startup of the playback is extremely slow. It is enabled by |
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351 default.</DD> |
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352 |
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353 <DT><CODE>type <0-2></CODE><DT> |
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354 <DD>is an optional integer between <CODE>0</CODE> and <CODE>2</CODE> that |
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355 selects which resampling method to use. Here <CODE>0</CODE> represents |
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356 linear interpolation as resampling method, <CODE>1</CODE> represents |
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357 resampling using a poly-phase filter-bank and integer processing and |
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358 <CODE>2</CODE> represents resampling using a poly-phase filter-bank and |
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359 floating point processing. Linear interpolation is extremely fast, but |
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360 suffers from poor sound quality especially when used for up-sampling. The |
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361 best quality is given by <CODE>2</CODE> but this method also suffers from |
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362 the highest CPU load.</DD> |
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363 </DL> |
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364 |
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365 <P>Example:<BR> |
8950 | 366 <CODE>mplayer -af resample=44100:0:0</CODE></P> |
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367 |
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368 <P>would set the output frequency of the resample filter to 44100Hz using exact |
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369 output frequency scaling and linear interpolation.</P> |
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370 |
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371 |
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372 <H5><A NAME="af_channels">2.3.2.3.2 Changing the number of channels</A></H5> |
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373 |
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374 <P>The <CODE>channels</CODE> filter can be used for adding and removing |
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375 channels, it can also be used for routing or copying channels. It is |
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376 automatically enabled when the output from the audio filter layer differs from |
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377 the input layer or when it is requested by another filter. This filter unloads |
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378 itself if not needed. The number of options is dynamic:</P> |
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379 |
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380 <DL> |
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381 <DT><CODE>nch <1-6></CODE></DT> |
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382 <DD>is an integer between <CODE>1</CODE> and <CODE>6</CODE> that is used for |
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383 setting the number of output channels. This option is required, leaving it |
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384 empty results in a runtime error.</DD> |
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385 |
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386 <DT><CODE>nr <1-6></CODE></DT> |
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387 <DD>is an integer between <CODE>1</CODE> and <CODE>6</CODE> that is used for |
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388 specifying the number of routes. This parameter is optional. If it is |
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389 omitted the default routing is used.</DD> |
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390 |
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391 <DT><CODE>from1:to1:from2:to2:from3:to3...</CODE></DT> |
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392 <DD>are pairs of numbers between <CODE>0</CODE> and <CODE>5</CODE> that define |
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393 where each channel should be routed.</DD> |
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394 </DL> |
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395 |
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396 <P>If only <CODE>nch</CODE> is given the default routing is used, it works as |
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397 follows: If the number of output channels is bigger than the number of input |
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398 channels empty channels are inserted (except mixing from mono to stereo, then |
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399 the mono channel is repeated in both of the output channels). If the number of |
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400 output channels is smaller than the number of input channels the exceeding |
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401 channels are truncated.</P> |
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402 |
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403 <P>Example 1:<BR> |
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404 <CODE>mplayer -af channels=4:4:0:1:1:0:2:2:3:3 media.avi </CODE></P> |
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405 |
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406 <P>would change the number of channels to 4 and set up 4 routes that swap |
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407 channel 0 and channel 1 and leave channel 2 and 3 intact. Observe that if |
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408 media containing two channels was played back, channels 2 and 3 would contain |
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409 silence but 0 and 1 would still be swapped.</P> |
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410 |
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411 <P>Example 2:<BR> |
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412 <CODE>mplayer -af channels=6:4:0:0:0:1:0:2:0:3 media.avi </CODE></P> |
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413 |
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414 <P>would change the number of channels to 6 and set up 4 routes that copy |
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415 channel 0 to channels 0 to 3. Channel 4 and 5 will contain silence.</P> |
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416 |
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417 |
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418 <H5><A NAME="af_format">2.3.2.3.3 Sample format converter</A></H5> |
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419 |
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421 is automatically enabled when needed by the sound card or another filter.</P> | |
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422 |
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423 <DL> |
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424 <DT><CODE>bps <number></CODE></DT> |
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425 <DD>can be <CODE>1</CODE>, <CODE>2</CODE> or <CODE>4</CODE> and denotes the |
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426 number of bytes per sample. This option is required, leaving it empty |
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427 results in a runtime error.</DD> |
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428 |
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429 <DT><CODE>f <format></CODE></DT> |
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430 <DD>is a text string describing the sample format. The string is a |
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431 concatenated mix of: <CODE>alaw</CODE>, <CODE>mulaw</CODE> or |
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432 <CODE>imaadpcm</CODE>, <CODE>float</CODE> or <CODE>int</CODE>, |
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433 <CODE>unsigned</CODE> or <CODE>signed</CODE>, <CODE>le</CODE> or |
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434 <CODE>be</CODE> (little or big endian). This option is required, leaving it |
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435 empty results in a runtime error.</DD> |
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436 </DL> |
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437 |
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438 <P>Example:<BR> |
8950 | 439 <CODE>mplayer -af format=4:float media.avi</CODE></P> |
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440 |
8772 | 441 <P>would set the output format to 4 bytes per sample floating point |
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442 data.</P> |
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443 |
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444 |
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445 <H5><A NAME="af_delay">2.3.2.3.4 Delay</A></H5> |
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446 |
8950 | 447 <P>The <CODE>delay</CODE> filter delays the sound to the loudspeakers such that |
448 the sound from the different channels arrives at the listening position | |
449 simultaneously. | |
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450 It is only useful if you have more than 2 loudspeakers. This filter has a |
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451 variable number of parameters:</P> |
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452 |
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453 <DL> |
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454 <DT><CODE>d1:d2:d3...</CODE></DT> |
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455 <DD>are floating point numbers representing the delays in ms that should be |
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456 imposed on the different channels. The minimum delay is 0ms and the maximum |
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457 is 1000ms.</DD> |
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458 </DL> |
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459 |
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460 <P>To calculate the required delay for the different channels do as follows:</P> |
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461 |
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462 <OL> |
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463 <LI>Measure the distance to the loudspeakers in meters in relation to your |
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464 listening position, giving you the distances s1 to s5 (for a 5.1 system). |
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465 There is no point in compensating for the sub-woofer (you will not hear the |
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466 difference anyway).</LI> |
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467 <LI>Subtract the distances s1 to s5 from the maximum distance i.e.<BR> |
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468 s[i] = max(s) - s[i]; i = 1...5</LI> |
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469 <LI>Calculated the required delays in ms as<BR> |
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470 d[i] = 1000*s[i]/342; i = 1...5 </LI> |
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471 </OL> |
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472 |
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473 <P>Example:<BR> |
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474 <CODE>mplayer -af delay=10.5:10.5:0:0:7:0 media.avi</CODE></P> |
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475 |
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476 <P>would delay front left and right by 10.5ms, the two rear channels and the sub |
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477 by 0ms and the center channel by 7ms.</P> |
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478 |
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479 |
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480 <H5><A NAME="af_volume">2.3.2.3.5 Software volume control</A></H5> |
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481 |
8950 | 482 <P>Software volume control is implemented by the <CODE>volume</CODE> audio |
483 filter. Use this filter with caution since | |
8750
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484 it can reduce the signal to noise ratio of the sound. In most cases it is best |
9166 | 485 to set the level for the MASTER sound to max, leave this filter out and control |
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486 the output level to your speakers with the master volume control of the mixer. |
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487 If there is an external amplifier connected to the computer (this is almost |
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488 always the case), the noise level can be minimized by adjusting the master |
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489 level and the volume knob on the amplifier until the hissing noise in the |
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490 background is gone. This filter has two options:</P> |
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491 |
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492 <DL> |
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493 <DT><CODE>v <-200 - +60></CODE></DT> |
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494 <DD>is a floating point number between <CODE>-200</CODE> and <CODE>+60</CODE> |
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495 which represents the volume level in dB. The default level is 0dB.</DD> |
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496 |
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497 <DT><CODE>c</CODE></DT> |
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498 <DD>is a binary control that turns soft clipping on and off. Soft-clipping can |
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499 make the sound more smooth if very high volume levels are used. Enable this |
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500 option if the dynamic range of the loudspeakers is very low. Be aware that |
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501 this feature creates distortion and should be considered a last resort.</DD> |
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502 </DL> |
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503 |
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504 <P>Example:<BR> |
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505 <CODE>mplayer -af volume=10.1:0 media.avi</CODE></P> |
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506 |
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507 <P>would amplify the sound by 10.1dB and hard-clip if the sound level is too |
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508 high.</P> |
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509 |
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510 <P>This filter has a second feature: It measures the overall maximum sound level |
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511 and prints out that level when MPlayer exits. This volume estimate can be used |
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512 for setting the sound level in MEncoder such that the maximum dynamic range is |
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513 utilized.</P> |
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514 |
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515 |
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516 <H5><A NAME="af_equalizer">2.3.2.3.6 Equalizer</A></H5> |
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517 |
8950 | 518 <P>The <CODE>equalizer</CODE> filter represents a 10 octave band graphic |
519 equalizer, implemented using 10 IIR | |
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520 band pass filters. This means that it works regardless of what type of audio |
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521 is being played back. The center frequencies for the 10 bands are:</P> |
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522 |
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|
523 <TABLE BORDER="0" WIDTH="100%"> |
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524 <TR><TD>Band No.</TD><TD>Center frequency</TD></TR> |
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525 <TR><TD>0</TD><TD>31.25 Hz</TD></TR> |
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526 <TR><TD>1</TD><TD>62.50 Hz</TD></TR> |
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527 <TR><TD>2</TD><TD>125.0 Hz</TD></TR> |
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528 <TR><TD>3</TD><TD>250.0 Hz</TD></TR> |
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529 <TR><TD>4</TD><TD>500.0 Hz</TD></TR> |
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530 <TR><TD>5</TD><TD>1.000 kHz</TD></TR> |
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531 <TR><TD>6</TD><TD>2.000 kHz</TD></TR> |
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532 <TR><TD>7</TD><TD>4.000 kHz</TD></TR> |
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533 <TR><TD>8</TD><TD>8.000 kHz</TD></TR> |
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534 <TR><TD>9</TD><TD>16.00 kHz</TD></TR> |
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535 </TABLE> |
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536 |
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537 <P>If the sample rate of the sound being played back is lower than the center |
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538 frequency for a frequency band, then that band will be disabled. A known bug |
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539 with this filter is that the characteristics for the uppermost band are not |
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New audio filter documentation by Anders Johannsson with some structural
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|
540 completely symmetric if the sample rate is close to the center frequency of |
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New audio filter documentation by Anders Johannsson with some structural
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|
541 that band. This problem can be worked around by up-sampling the sound using |
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New audio filter documentation by Anders Johannsson with some structural
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|
542 the resample filter before it reaches this filter. </P> |
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New audio filter documentation by Anders Johannsson with some structural
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|
543 |
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544 <P>This filter has 10 parameters:</P> |
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|
545 |
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|
546 <DL> |
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|
547 <DT><CODE>g1:g2:g3...g10</CODE></DT> |
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Some updates and rewordings as well as a much better table design by
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|
548 <DD>are floating point numbers between <CODE>-12</CODE> and <CODE>+12</CODE> |
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Further libaf documentation by Anders with some more updates by me.
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549 representing the gain in dB for each frequency band.</DD> |
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New audio filter documentation by Anders Johannsson with some structural
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|
550 </DL> |
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New audio filter documentation by Anders Johannsson with some structural
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551 |
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|
552 <P>Example:<BR> |
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|
553 <CODE>mplayer -af equalizer=11:11:10:5:0:-12:0:5:12:12 media.avi</CODE></P> |
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New audio filter documentation by Anders Johannsson with some structural
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|
554 |
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|
555 <P>would amplify the sound in the upper and lower frequency region while |
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New audio filter documentation by Anders Johannsson with some structural
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|
556 canceling it almost completely around 1kHz.</P> |
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557 |
8950 | 558 |
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559 <H5><A NAME="af_panning">2.3.2.3.7 Panning filter</A></H5> |
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|
560 |
8950 | 561 <P>Use the <CODE>pan</CODE> filter to mix channels arbitrarily. It is basically |
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562 a combination of the volume control and the channels filter. There are two |
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563 major uses for this filter:</P> |
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|
564 |
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|
565 <OL> |
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New audio filter documentation by Anders Johannsson with some structural
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|
566 <LI>Down-mixing many channels to only a few, stereo to mono for example.</LI> |
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New audio filter documentation by Anders Johannsson with some structural
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|
567 <LI>Varying the "width" of the center speaker in a surround sound system.</LI> |
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|
568 </OL> |
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New audio filter documentation by Anders Johannsson with some structural
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|
569 |
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New audio filter documentation by Anders Johannsson with some structural
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|
570 <P>This filter is hard to use, and will require some tinkering before the |
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571 desired result is obtained. The number of options for this filter depends on |
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|
572 the number of output channels:</P> |
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|
573 |
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|
574 <DL> |
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Further libaf documentation by Anders with some more updates by me.
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575 <DT><CODE>nch <1-6></CODE></DT> |
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576 <DD>is an integer between <CODE>1</CODE> and <CODE>6</CODE> and is used for |
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577 setting the number of output channels. This option is required, leaving it |
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|
578 empty results in a runtime error.</DD> |
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|
579 |
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|
580 <DT><CODE>l00:l01:l02:..l10:l11:l12:...ln0:ln1:ln2:...</CODE></DT> |
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Further libaf documentation by Anders with some more updates by me.
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|
581 <DD>are floating point values between <CODE>0</CODE> and <CODE>1</CODE>. |
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Further libaf documentation by Anders with some more updates by me.
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|
582 <CODE>l[i][j]</CODE> determines how much of input channel j is mixed into |
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Further libaf documentation by Anders with some more updates by me.
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|
583 output channel i.</DD> |
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New audio filter documentation by Anders Johannsson with some structural
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|
584 </DL> |
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585 |
8849 | 586 <P>Example 1:<BR> |
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587 <CODE>mplayer -af pan=1:0.5:0.5 -channels 1 media.avi</CODE></P> |
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588 |
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589 <P>would down-mix from stereo to mono.</P> |
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590 |
8849 | 591 <P>Example 2:<BR> |
592 <CODE>mplayer -af pan=3:1:0:1:0.5:0.5 -channels 3 media.avi</CODE></P> | |
593 | |
594 <P>would give 3 channel output leaving channels 0 and 1 intact, and mix channels | |
595 0 and 1 into output channel 2 (which could be sent to a sub-woofer for | |
596 example).</P> | |
597 | |
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598 |
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599 <H5><A NAME="af_sub">2.3.2.3.8 Sub-woofer</A></H5> |
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600 |
8950 | 601 <P>The <CODE>sub</CODE> filter adds a sub woofer channel to the audio stream. |
602 The audio data | |
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603 used for creating the sub-woofer channel is an average of the sound in channel |
9005 | 604 0 and channel 1. The resulting sound is then low-pass filtered by a 4th |
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605 order Butterworth filter with a default cutoff frequency of 60Hz and added to |
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606 a separate channel in the audio stream. Warning: Disable this filter when you |
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607 are playing DVDs with Dolby Digital 5.1 sound, otherwise this filter will |
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608 disrupt the sound to the sub-woofer. This filter has two parameters:</P> |
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609 |
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610 <DL> |
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611 <DT><CODE>fc <20-300></CODE></DT> |
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612 <DD>is an optional floating point number used for setting the cutoff frequency |
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613 for the filter in Hz. The valid range is 20Hz to 300Hz. For the best result |
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614 try setting the cutoff frequency as low as possible. This will improve the |
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615 stereo or surround sound experience. The default cutoff frequency is |
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616 60Hz.</DD> |
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617 |
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618 <DT><CODE>ch <0-5></CODE></DT> |
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619 <DD>is an optional integer between <CODE>0</CODE> and <CODE>5</CODE> which |
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620 determines the channel number in which to insert the sub-channel audio. |
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621 The default is channel number <CODE>5</CODE>. Observe that the number of |
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622 channels will automatically be increased to <CODE>ch</CODE> if |
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623 necessary.</DD> |
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624 </DL> |
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625 |
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626 <P>Example:<BR> |
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627 <CODE>mplayer -af sub=100:4 -channels 5 media.avi</CODE></P> |
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628 |
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629 <P>would add a sub-woofer channel with a cutoff frequency of 100Hz to output |
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630 channel 4.</P> |
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631 |
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632 <H5><A NAME="af_surround">2.3.2.3.9 Surround-sound decoder</A></H5> |
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633 |
8950 | 634 <P>Matrix encoded surround sound can be decoded by the <CODE>surround</CODE> |
635 filter. Dolby Surround is | |
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636 an example of a matrix encoded format. Many files with 2 channel audio |
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637 actually contain matrixed surround sound. To use this feature you need a sound |
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638 card supporting at least 4 channels. This filter has one parameter:</P> |
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639 |
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640 <DL> |
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641 <DT><CODE>d <0-1000></CODE></DT> |
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642 <DD>is an optional floating point number between <CODE>0</CODE> and |
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643 <CODE>1000</CODE> used for setting the delay time in ms for the rear |
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644 speakers. This delay should be set as follows: if d1 is the distance from |
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645 the listening position to the front speakers and d2 is the distance from |
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646 the listening position to the rear speakers, then the delay <CODE>d</CODE> |
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647 should be set to 15ms if d1 <= d2 and to 15 + 5*(d1-d2) if d1 > d2. |
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648 The default value for <CODE>d</CODE> is 20ms.</DD> |
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649 </DL> |
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650 |
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651 <P>Example:<BR> |
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652 <CODE>mplayer -af surround=15 -channels 4 media.avi</CODE></P> |
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653 |
8950 | 654 <P>would add surround sound decoding with 15ms delay for the sound to the rear |
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655 speakers.</P> |
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656 |
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657 |
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658 <H4><A NAME="plugins">2.3.2.4 Audio plugins (deprecated)</A></H4> |
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659 |
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660 <H2><STRONG>Note: Audio plugins have been deprecated by audio filters and will be |
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661 removed soon.</STRONG></H2> |
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662 |
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663 <P>MPlayer has support for audio plugins. Audio plugins can be used to |
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664 change the properties of the audio data before it reaches the sound |
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665 card. They are enabled using the <CODE>-aop</CODE> option which takes a |
5765 | 666 <CODE>list=plugin1,plugin2,...</CODE> argument. The <CODE>list</CODE> argument |
667 is required and determines which plugins should be used and in which order they | |
6974 | 668 should be executed. Example:</P> |
4508 | 669 |
670 <P> <CODE>mplayer media.avi -aop list=resample,format</CODE></P> | |
671 | |
672 <P>would run the sound through the resampling plugin followed by the format | |
6974 | 673 plugin.</P> |
4508 | 674 |
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675 <P>The plugins can also have options that change their behavior. These |
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676 options are explained in detail in the sections below. A plugin will execute |
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677 using default settings if its options are omitted. Here is an example of how |
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678 to use plugins in combination with plugin specific options:</P> |
4508 | 679 |
680 <P> <CODE>mplayer media.avi -aop | |
6974 | 681 list=resample,format:fout=44100:format=0x8</CODE></P> |
4508 | 682 |
683 <P>would set the output frequency of the resample plugin to 44100Hz and the | |
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684 output format of the format plugin to AFMT_U8.</P> |
4508 | 685 |
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686 <P>Currently audio plugins cannot be used in MEncoder.</P> |
4508 | 687 |
688 | |
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689 <H5><A NAME="resample">2.3.2.4.1 Up/Downsampling</A></H5> |
3671 | 690 |
7814 | 691 <P>MPlayer fully supports up/downsampling of the sound. This plugin can |
5765 | 692 be used if you have a fixed frequency sound card or if you are |
693 stuck with an old sound card that is only capable of max 44.1kHz. | |
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694 MPlayer <EM>autodetects</EM> whether or not usage of this plugin is necessary. |
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695 This plugin has one option, <CODE>fout</CODE>, which is used for setting the |
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696 desired output sample frequency. The value is given in Hz, and defaults to |
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697 48kHz.</P> |
3671 | 698 |
5765 | 699 <P>Usage:<BR> |
6974 | 700 <CODE>mplayer media.avi -aop list=resample:fout=<required |
3671 | 701 frequency in Hz, like 44100></CODE></P> |
702 | |
5765 | 703 <P>Note that the output frequency should not be scaled up from the default value. |
704 Scaling up will cause the audio and video streams to be played in slow motion | |
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705 and cause audio distortion.</P> |
3671 | 706 |
6974 | 707 |
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708 <H5><A NAME="surround_decoding">2.3.2.4.2 Surround Sound decoding</A></H5> |
3931 | 709 |
7814 | 710 <P>MPlayer has an audio plugin that can decode matrix encoded |
6974 | 711 surround sound. Dolby Surround is an example of a matrix encoded format. |
712 Many files with 2 channel audio actually contain matrixed surround sound. | |
713 To use this feature you need a sound card supporting at least 4 channels.</P> | |
3931 | 714 |
5765 | 715 <P>Usage:<BR> |
6974 | 716 <CODE>mplayer media.avi -aop list=surround</CODE></P> |
3931 | 717 |
4508 | 718 |
8177 | 719 <H5><A NAME="format">2.3.2.3.3 Sample format converter</A></H5> |
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720 |
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721 <P>If your sound card driver does not support signed 16bit <CODE>int</CODE> data type, |
5765 | 722 this plugin can |
4508 | 723 be used to change the format to one which your sound card can understand. It |
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724 has one option, <CODE>format</CODE>, which can be set to one of the numbers |
5765 | 725 found in <CODE>libao2/afmt.h</CODE>. This plugin is hardly ever needed and is |
726 intended for advanced users. Keep in mind that this plugin only changes the | |
6974 | 727 sample format and not the sample frequency or the number of channels.</P> |
4508 | 728 |
5765 | 729 <P>Usage:<BR> |
4508 | 730 <CODE>mplayer media.avi -aop |
6974 | 731 list=format:format=<required output format></CODE></P> |
4508 | 732 |
733 | |
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734 <H5><A NAME="delay">2.3.2.4.4 Delay</A></H5> |
4508 | 735 |
736 <P>This plugin delays the sound and is intended as an example of how to develop | |
5765 | 737 new plugins. It can not be used for anything useful from a users perspective |
4508 | 738 and is mentioned here for the sake of completeness only. Do not use this |
739 plugin unless you are a developer.</P> | |
740 | |
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741 <P>If you have a file with a consistent A/V sync fault, use the <CODE>+/-</CODE> |
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742 keys to adjust timings on-the-fly instead. Usage of the OSD is recommended |
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743 to make this easier.</P> |
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744 |
7047 | 745 |
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746 <H5><A NAME="volume">2.3.2.4.5 Software volume control</A></H5> |
4913 | 747 |
748 <P>This plugin is a software replacement for the volume control, and | |
5765 | 749 can be used on machines with a broken mixer device. It can also be |
7814 | 750 used if one wants to change the output volume of MPlayer |
4913 | 751 without changing the PCM volume setting in the mixer. It has one |
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752 option <CODE>volume</CODE> that is used for setting the initial |
4913 | 753 sound level. The initial sound level can be set to values between 0 |
5099 | 754 and 255 and defaults to 101 which equals 0dB amplification. Use this |
755 plugin with caution since it can reduce the signal to noise ratio of | |
9166 | 756 the sound. In most cases it is best to set the level for the MASTER |
5099 | 757 sound to max, leave this plugin out and control the output level to |
5765 | 758 your speakers with the master volume control of the mixer. If there is an |
5099 | 759 external amplifier connected to the computer (this is almost always |
760 the case), the noise level can be minimized by adjusting the master | |
761 level and the volume knob on the amplifier until the hissing noise | |
6974 | 762 in the background is gone.</P> |
4913 | 763 |
5765 | 764 <P>Usage:<BR> |
4913 | 765 <CODE>mplayer media.avi -aop |
6974 | 766 list=volume:volume=<0-255></CODE></P> |
4913 | 767 |
5765 | 768 <P>This plugin also has compressor or "soft-clipping" capabilities. |
5099 | 769 Compression can be used if the dynamic range of the sound is very |
770 high or if the dynamic range of the loudspeakers is very | |
5765 | 771 low. Be aware that this feature creates distortion and should be |
6974 | 772 considered a last resort.</P> |
5099 | 773 |
5765 | 774 <P>Usage:<BR> |
5099 | 775 <CODE>mplayer media.avi -aop |
6974 | 776 list=volume:softclip</CODE></P> |
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777 |
5109 | 778 |
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779 <H5><A NAME="extrastereo">2.3.2.4.6 Extrastereo</A></H5> |
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780 |
5765 | 781 <P>This plugin (linearly) increases the difference between left and right |
782 channels (like the XMMS extrastereo plugin) which gives some sort of "live" | |
6974 | 783 effect to playback.</P> |
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784 |
5765 | 785 <P>Usage:<BR> |
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786 <CODE>mplayer media.avi -aop list=extrastereo</CODE><BR> |
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787 <CODE>mplayer media.avi -aop list=extrastereo:mul=3.45</CODE></P> |
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788 |
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added documentation for "extrastereo" plugin (patch by pl)
gabucino
parents:
4913
diff
changeset
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789 <P>The default coefficient (<CODE>mul</CODE>) is a float number that defaults |
5765 | 790 to 2.5. If you set it to 0.0, you will have mono sound (average of both |
6272 | 791 channels). If you set it to 1.0, sound will be unchanged, if you set it to |
792 -1.0, left and right channels will be swapped.</P> | |
4928
ecf9f93dd9b6
added documentation for "extrastereo" plugin (patch by pl)
gabucino
parents:
4913
diff
changeset
|
793 |
5109 | 794 |
8750
f6323ff433aa
New audio filter documentation by Anders Johannsson with some structural
diego
parents:
8260
diff
changeset
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795 <H5><A NAME="normalizer">2.3.2.4.7 Volume normalizer</A></H5> |
5109 | 796 |
5765 | 797 <P>This plugin maximizes the volume without distorting the sound.</P> |
5109 | 798 |
5765 | 799 <P>Usage:<BR> |
5109 | 800 <CODE>mplayer media.avi -aop list=volnorm</CODE><BR> |
801 | |
5721 | 802 |
1612 | 803 </BODY> |
804 </HTML> |