Mercurial > mplayer.hg
annotate DOCS/sound.html @ 8768:44b7d4c7e61c
cosmetics
author | attila |
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date | Sat, 04 Jan 2003 13:12:13 +0000 |
parents | 52088761d77e |
children | 605f2c290b20 |
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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> | |
25 <DD>SDL driver (supports <B>ESD</B>, <B>ARTS</B> etc)</DD> | |
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 |
8177 | 46 <P>Fact is, Linux sound card drivers have compatibility problems. The cause |
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47 is that MPlayer uses a feature that well coded audio drivers implement to |
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48 maintain audio/video sync. Regrettably, some driver authors do not care about |
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49 this function, it is not needed for playing MP3s or for sound effects.</P> |
8177 | 50 |
51 <P>Other media players like aviplay or xine possibly work out-of-the-box with | |
52 these drivers because they use "simple" methods with internal timing. A note: | |
53 time showed their methods aren't AS efficient as MPlayer's.</P> | |
54 | |
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55 <P>With a correctly written audio driver MPlayer will never create audio related |
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56 A/V desynchronisation, unless your file is badly broken. Some options to work |
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57 around these problems are described in the man page).</P> |
8177 | 58 |
59 <P>If you happen to have a bad audio driver, try the <CODE>-autosync</CODE> | |
60 option, it should sort out your problems. See the man page for detailed | |
61 information.</P> | |
62 | |
63 <P>Some notes:</P> | |
5765 | 64 |
65 <UL> | |
6960 | 66 <LI>If you have an OSS driver, first try <CODE>-ao oss</CODE> (this is the |
67 default). If you experience glitches, halts or anything out of the | |
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68 ordinary, try <CODE>-ao sdl</CODE> (NOTE: You need to have SDL libraries |
6960 | 69 and header files installed). The SDL audio driver helps in a lot of cases |
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70 and also supports ESD and ARTS. (ESD is the sound daemon |
6960 | 71 from GNOME, ARTS is from KDE.)</LI> |
72 <LI>If you have ALSA version 0.5, then you almost always have to use | |
73 <CODE>-ao alsa5</CODE> , since ALSA 0.5 has buggy OSS emulation code, and | |
74 will <B>crash MPlayer</B> with a message like this:<BR> | |
75 <CODE>DEMUXER: Too many (945 in 8390980 bytes) video packets in the buffer!</CODE></LI> | |
5765 | 76 </UL> |
1892 | 77 |
8177 | 78 <P>On <B>Solaris</B>, use the SUN audio driver with the |
6960 | 79 <CODE>-ao sun</CODE> option, otherwise neither video nor audio will work.</P> |
1892 | 80 |
7047 | 81 |
8177 | 82 <H4><A NAME="experiences">2.3.2.2 Sound Card experiences, recommendations</A></H4> |
1612 | 83 |
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84 <TABLE BORDER="0" WIDTH="100%"> |
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85 <TR><TD COLSPAN=3><B>VIA onboard chipset (via82cxxx) 48kHz only</B></TD></TR> |
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86 <TR><TD></TD><TD>Driver:</TD><TD> from the |
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87 <A HREF="http://sourceforge.net/project/showfiles.php?group_id=3242&release_id=59602">gkernel project</A></TD></TR> |
3190 | 88 |
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89 <TR><TD COLSPAN=3><B>Aureal Vortex 2</B></TD></TR> |
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90 <TR><TD> </TD><TD>OSS:</TD><TD>no driver</TD></TR> |
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91 <TR><TD></TD><TD>OSS/Pro:</TD><TD>OK</TD></TR> |
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92 <TR><TD></TD><TD>ALSA:</TD><TD>no driver</TD></TR> |
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93 <TR><TD></TD><TD>Max kHz:</TD><TD>48</TD></TR> |
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94 <TR><TD></TD><TD>Driver:</TD><TD><A HREF="http://aureal.sourceforge.net">aureal.sourceforge.net</A></TD></TR> |
8751 | 95 <TR><TD></TD><TD>Driver2:</TD><TD> from <A HREF="http://makacs.poliod.hu/~pontscho/aureal/au88xx-1.1.3.tar.bz2">Pontscho's page</A> |
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96 (<I>buffer size increased to 32k</I>)</TD></TR> |
1612 | 97 |
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98 <TR><TD COLSPAN=3><B>GUS PnP</B></TD></TR> |
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99 <TR><TD></TD><TD>OSS:</TD><TD>no driver</TD></TR> |
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100 <TR><TD></TD><TD>OSS/Pro:</TD><TD>OK</TD></TR> |
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101 <TR><TD></TD><TD>ALSA:</TD><TD>OK</TD></TR> |
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102 <TR><TD></TD><TD>Max kHz:</TD><TD>48</TD></TR> |
1682 | 103 |
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104 <TR><TD COLSPAN=3><B>SB Live!</B></TD></TR> |
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105 <TR><TD></TD><TD>OSS:</TD><TD>Analog OK, SP/DIF not working</TD></TR> |
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106 <TR><TD></TD><TD>ALSA:</TD><TD>Both OK</TD></TR> |
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107 <TR><TD></TD><TD>Max kHz:</TD><TD>192</TD></TR> |
1612 | 108 |
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109 <TR><TD COLSPAN=3><B>SB AWE 64</B></TD></TR> |
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110 <TR><TD></TD><TD>OSS:</TD><TD>max 44kHz</TD></TR> |
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111 <TR><TD></TD><TD>ALSA:</TD><TD>48kHz sounds bad</TD></TR> |
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112 <TR><TD></TD><TD>Max kHz:</TD><TD>48</TD></TR> |
1892 | 113 |
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114 <TR><TD COLSPAN=3><B>Gravis UltraSound ACE</B></TD></TR> |
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115 <TR><TD></TD><TD>OSS:</TD><TD>not OK</TD></TR> |
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116 <TR><TD></TD><TD>ALSA:</TD><TD>OK</TD></TR> |
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117 <TR><TD></TD><TD>Max kHz:</TD><TD>44</TD></TR> |
1612 | 118 |
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119 <TR><TD COLSPAN=3><B>Gravis UltraSound MAX</B></TD></TR> |
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120 <TR><TD></TD><TD>OSS:</TD><TD>OK</TD></TR> |
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121 <TR><TD></TD><TD>ALSA:</TD><TD>OK (?)</TD></TR> |
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122 <TR><TD></TD><TD>Max kHz:</TD><TD>48</TD></TR> |
1612 | 123 |
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124 <TR><TD COLSPAN=3><B>ESS 688</B></TD></TR> |
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125 <TR><TD></TD><TD>OSS:</TD><TD>OK</TD></TR> |
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126 <TR><TD></TD><TD>ALSA:</TD><TD>OK (?)</TD></TR> |
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127 <TR><TD></TD><TD>Max kHz:</TD><TD>48</TD></TR> |
1612 | 128 |
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129 <TR><TD COLSPAN=3><B>C-Media cards (which ones?)</B></TD></TR> |
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130 <TR><TD></TD><TD>OSS:</TD><TD>not OK (hissing) (?)</TD></TR> |
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131 <TR><TD></TD><TD>ALSA:</TD><TD>OK (?)</TD></TR> |
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132 <TR><TD></TD><TD>Max kHz:</TD><TD>?</TD></TR> |
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134 <TR><TD COLSPAN=3><B>Yamaha cards (*ymf*)</B></TD></TR> |
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135 <TR><TD></TD><TD>OSS:</TD><TD>not OK (?) (maybe <CODE>-ao sdl</CODE>)</TD></TR> |
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136 <TR><TD></TD><TD>ALSA:</TD><TD>OK only with ALSA 0.5 with OSS emulation <B>AND</B> <CODE>-ao sdl</CODE> (!) (?)</TD></TR> |
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137 <TR><TD></TD><TD>Max kHz:</TD><TD>?</TD></TR> |
1612 | 138 |
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139 <TR><TD COLSPAN=3><B>Cards with envy24 chips (like Terratec EWS88MT)</B></TD></TR> |
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140 <TR><TD></TD><TD>OSS:</TD><TD>?</TD></TR> |
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141 <TR><TD></TD><TD>OSS/Pro:</TD><TD>OK</TD></TR> |
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142 <TR><TD></TD><TD>ALSA:</TD><TD>?</TD></TR> |
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143 <TR><TD></TD><TD>Max kHz:</TD><TD>?</TD></TR> |
3438 | 144 |
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145 <TR><TD COLSPAN=3><B>PC Speaker or DAC</B></TD></TR> |
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146 <TR><TD></TD><TD>OSS:</TD><TD>OK (Use the SDL driver: <CODE>-ao sdl</CODE>)</TD></TR> |
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147 <TR><TD></TD><TD>ALSA:</TD><TD>no driver</TD></TR> |
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148 <TR><TD></TD><TD>Max kHz:</TD><TD>The driver emulates 44.1, maybe more.</TD></TR> |
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149 <TR><TD></TD><TD>Driver:</TD><TD><A HREF="ftp://ftp.infradead.org/pub/pcsp">ftp://ftp.infradead.org/pub/pcsp</A></TD></TR> |
5765 | 150 </TABLE> |
1612 | 151 |
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152 <P>On Linux, a 2.4.x kernel is highly recommended. Kernel 2.2 is not tested.</P> |
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153 |
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154 <P>If the sound clicks when playing from CD-ROM, turn on IRQ unmasking, e.g. |
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155 <CODE>hdparm -u1 /dev/cdrom</CODE> (<CODE>man hdparm</CODE>). This is |
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156 generally beneficial and described in more detail in the |
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157 <A HREF="cd-dvd.html#drives">CD-ROM section</A>.</P> |
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158 |
7814 | 159 <P>Feedback to this document is welcome. Please tell us how MPlayer |
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160 and your sound card(s) worked together.</P> |
1612 | 161 |
3671 | 162 |
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163 <H4><A NAME="af">2.3.2.3 Audio filters</A></H4> |
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164 |
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165 <P>The old audio plugins have been superseded by a new audio filter layer. Audio |
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166 filters are used for changing the properties of the audio data before the |
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167 sound reaches the sound card. The activation and deactivation of the filters |
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168 is normally automated but can be overridden. The filters are activated when |
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169 the properties of the audio data differ from those required by the sound card |
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170 and deactivated if unnecessary. The <CODE>-af filter1,filter2,...</CODE> |
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171 switch is used to override the automatic activation of filters or to insert |
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172 filters that are not automatically inserted. The filters will be executed as |
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173 they appear in the comma separated list.</P> |
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174 |
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175 <P>Example:<BR> |
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176 <CODE>mplayer -af resample,pan movie.avi </CODE></P> |
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177 |
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178 <P>would run the sound through the resampling filter followed by the pan filter. |
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179 Observe that the list must not contain any spaces, else it will fail.</P> |
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180 |
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181 <P>The filters often have switches that change their behavior. These switches |
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182 are explained in detail in the sections below. A filter will execute using |
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183 default settings if its switches are omitted. Here is an example of how to use |
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184 filters in combination with filter specific switches:</P> |
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185 |
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186 <P> <CODE>mplayer -af resample=11025,pan=1:0.5:0.5 -channels 1 |
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187 -srate 11025 media.avi</CODE></P> |
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188 |
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189 <P>would set the output frequency of the resample filter to 11025Hz and downmix |
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190 the audio to 1 channel using the pan filter.</P> |
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191 |
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192 <P>Most filters respond to the <CODE>-v</CODE> switch, which makes the filters |
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193 print out status messages.</P> |
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194 |
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195 <P>The overall execution of the filter layer is controlled using the |
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196 <CODE>-af-adv</CODE> switch. This switch has two suboptions:</P> |
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197 |
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198 <DL> |
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199 <DT><CODE>force</CODE><DT> |
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200 <DD>is an integer between 0 and 3 that controls how the filters are inserted |
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201 and what speed/accuracy optimizations they use: |
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202 <DL> |
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203 <DT>0</DT> |
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204 <DD>Use automatic insertion of filters and optimize according to CPU |
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205 speed.</DD> |
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206 <DT>1</DT> |
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207 <DD>Use automatic insertion of filters and optimize for the highest speed. |
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208 If this option is set the processing of the audio data will be done |
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209 using fix point arithmetics. Warning: Some features in the audio filters |
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210 will silently fail, and the sound quality may drop.</DD> |
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211 <DT>2</DT> |
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212 <DD>Use automatic insertion of filters and optimize for quality. If this |
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213 option is set the processing of the audio data will be done using |
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214 floating point instructions and is therefore quite CPU intensive, but |
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215 gives a lot higher sound quality than fix point processing.</DD> |
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216 <DT>3</DT> |
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217 <DD>Use no automatic insertion of filters and no optimization. Warning: It |
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218 may be possible to crash MPlayer using this setting.</DD> |
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219 </DL> |
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220 </DD> |
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221 |
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222 <DT><CODE>list</CODE></DT> |
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223 <DD>is an alias for the -af switch.</DD> |
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224 </DL> |
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225 |
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226 |
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227 <H5><A NAME="af_resample">2.3.2.3.1 Up/Down-sampling</A></H5> |
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228 |
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229 <P>MPlayer fully supports sound up/down-sampling. This filter can be used if you |
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230 have a fixed frequency sound card or if you are stuck with an old sound card |
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231 that is only capable of max 44.1kHz. This filter is automatically enabled if |
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232 it is necessary, but it can also be explicitly enabled on the command line. It |
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233 has three switches:</P> |
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234 |
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235 <DL> |
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236 <DT><CODE>srate</CODE></DT> |
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237 <DD>is an integer used for setting the output sample |
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238 frequency in Hz. The valid range for this parameter is 8kHz to 192kHz. If |
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239 the input and output sample frequency are the same or if this parameter is |
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240 omitted the filter is automatically unloaded. A high sample frequency |
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241 normally improves the audio quality, especially when used in combination |
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242 with other filters.</DD> |
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243 |
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244 <DT><CODE>sloppy</CODE></DT> |
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245 <DD>is an optional binary parameter that allows the output frequency to differ |
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246 slightly from the frequency given by <CODE>srate</CODE>. This switch can be |
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247 used if the startup of the playback is extremely slow.</DD> |
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248 |
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249 <DT><CODE>fast</CODE><DT> |
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250 <DD>is an optional binary parameter that enables linear interpolation as |
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251 resampling method. Linear interpolation is extremely fast, but suffers from |
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252 poor sound quality especially when used for up-sampling.</DD> |
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253 </DL> |
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254 |
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255 <P>Example:<BR> |
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256 <CODE>mplayer -af resample=44100:0:1</CODE></P> |
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257 |
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258 <P>would set the output frequency of the resample filter to 44100Hz using exact |
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259 output frequency scaling and linear interpolation.</P> |
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260 |
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261 |
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262 <H5><A NAME="af_channels">2.3.2.3.2 Changing the number of channels</A></H5> |
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263 |
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264 <P>The <CODE>channels</CODE> filter can be used for adding and removing |
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265 channels, it can also be used for routing or copying channels. It is |
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266 automatically enabled when the output from the audio filter layer differs from |
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267 the input layer or when it is requested by another filter. This filter unloads |
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268 itself if not needed. The number of switches is dynamic:</P> |
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269 |
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270 <DL> |
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271 <DT><CODE>nch</CODE></DT> |
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272 <DD>is an integer between 1 and 6 that is used for setting the number of |
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273 output channels. This switch is required, leaving it empty results in a |
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274 runtime error.</DD> |
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275 |
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276 <DT><CODE>nr</CODE></DT> |
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277 <DD>is an integer between 1 and 6 that is used for specifying the number of |
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278 routes. This parameter is optional. If it is omitted the default routing is |
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279 used.</DD> |
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280 |
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281 <DT><CODE>from1:to1:from2:to2:from3:to3...</CODE></DT> |
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282 <DD>are pairs of numbers between 0 and 5 that define where each channel should |
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283 be routed.</DD> |
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284 </DL> |
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285 |
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286 <P>If only <CODE>nch</CODE> is given the default routing is used, it works as |
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287 follows: If the number of output channels is bigger than the number of input |
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288 channels empty channels are inserted (except mixing from mono to stereo, then |
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289 the mono channel is repeated in both of the output channels). If the number of |
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290 output channels is smaller than the number of input channels the exceeding |
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291 channels are truncated.</P> |
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292 |
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293 <P>Example 1:<BR> |
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294 <CODE>mplayer -af channels=4:4:0:1:1:0:2:2:3:3 media.avi </CODE></P> |
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295 |
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296 <P>would change the number of channels to 4 and set up 4 routes that swap |
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297 channel 0 and channel 1 and leave channel 2 and 3 intact. Observe that if |
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298 media containing two channels was played back, channels 2 and 3 would contain |
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299 silence but 0 and 1 would still be swapped.</P> |
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300 |
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301 <P>Example 2:<BR> |
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302 <CODE>mplayer -af channels=6:4:0:0:0:1:0:2:0:3 media.avi </CODE></P> |
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303 |
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304 <P>would change the number of channels to 6 and set up 4 routes that copy |
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305 channel 0 to channels 0 to 3. Channel 4 and 5 will contain silence.</P> |
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306 |
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307 |
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308 <H5><A NAME="af_format">2.3.2.3.3 Sample format converter</A></H5> |
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309 |
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310 <P>This filter is a sample format converter. It is automatically enabled when |
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311 needed by the sound card or another filter.</P> |
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312 |
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313 <DL> |
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314 <DT><CODE>bps</CODE></DT> |
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315 <DD>can be 1, 2 or 4 and denotes the number of bytes per sample. This switch |
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316 is required, leaving it empty results in a runtime error.</DD> |
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317 |
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318 <DT><CODE>f</CODE></DT> |
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319 <DD>is a text string describing the sample format. The string is a |
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320 concatenated mix of: <CODE>alaw</CODE>, <CODE>mulaw</CODE> or |
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321 <CODE>imaadpcm</CODE>, <CODE>float</CODE> or <CODE>int</CODE>, |
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322 <CODE>unsigned</CODE> or <CODE>signed</CODE>, <CODE>le</CODE> or |
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323 <CODE>be</CODE> (little or big endian). This switch is required, leaving it |
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324 empty results in a runtime error.</DD> |
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325 </DL> |
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326 |
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327 <P>Example:<BR> |
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328 <CODE>mplayer media.avi -af format=4:float</CODE></P> |
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329 |
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330 <P>would set the output output format to 4 bytes per sample floating point |
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331 data.</P> |
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332 |
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333 |
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334 <H5><A NAME="af_delay">2.3.2.3.4 Delay</A></H5> |
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335 |
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336 <P>This filter delays the sound to the loudspeakers in order to make the sound |
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337 in the different channels arrive at the same time to the listening position. |
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338 It is only useful if you have more than 2 loudspeakers. This filter has a |
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339 variable number of parameters:</P> |
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340 |
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341 <DL> |
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342 <DT><CODE>d1:d2:d3...</CODE></DT> |
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343 <DD>are floating point numbers representing the delays in ms that should be |
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344 imposed on the different channels. The minimum delay is 0ms and the maximum |
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345 is 1000ms.</DD> |
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346 </DL> |
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347 |
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348 <P>To calculate the required delay for the different channels do as follows:</P> |
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349 |
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350 <OL> |
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351 <LI>Measure the distance to the loudspeakers in meters in relation to your |
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352 listening position, giving you the distances s1 to s5 (for a 5.1 system). |
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353 There is no point in compensating for the sub-woofer (you will not hear the |
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354 difference anyway).</LI> |
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355 <LI>Subtract the distances s1 to s5 from the maximum distance i.e.<BR> |
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356 s[i] = max(s) - s[i]; i = 1...5</LI> |
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357 <LI>Calculated the required delays in ms as<BR> |
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358 d[i] = 1000*s[i]/342; i = 1...5 </LI> |
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359 </OL> |
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360 |
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361 <P>Example:<BR> |
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362 <CODE>mplayer -af delay=10.5:10.5:0:0:7:0 media.avi</CODE></P> |
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363 |
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364 <P>would delay front left and right by 10.5ms, the two rear channels and the sub |
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365 by 0ms and the center channel by 7ms.</P> |
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366 |
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367 |
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368 <H5><A NAME="af_volume">2.3.2.3.5 Software volume control</A></H5> |
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369 |
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370 <P>This filter is a software volume control. Use this filter with caution since |
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371 it can reduce the signal to noise ratio of the sound. In most cases it is best |
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372 to set the level for the PCM sound to max, leave this filter out and control |
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373 the output level to your speakers with the master volume control of the mixer. |
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374 If there is an external amplifier connected to the computer (this is almost |
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375 always the case), the noise level can be minimized by adjusting the master |
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376 level and the volume knob on the amplifier until the hissing noise in the |
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377 background is gone. This filter has two switches:</P> |
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378 |
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379 <DL> |
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380 <DT><CODE>v</CODE></DT> |
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381 <DD>is a floating point number between -200 and +60 which represents the |
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382 volume level in dB. The default level is -10dB.</DD> |
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383 |
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384 <DT><CODE>c</CODE></DT> |
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385 <DD>is a binary control that turns soft clipping on and off. Soft-clipping can |
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386 make the sound more smooth if very high volume levels are used. Enable this |
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387 switch if the dynamic range of the loudspeakers is very low. Be aware that |
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388 this feature creates distortion and should be considered a last resort.</DD> |
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389 </DL> |
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390 |
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391 <P>Example:<BR> |
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392 <CODE>mplayer -af volume=10.1:0 media.avi</CODE></P> |
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393 |
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394 <P>would amplify the sound by 10.1dB and hard-clip if the sound level is too |
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395 high.</P> |
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396 |
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397 <P>This filter has a second feature: It measures the overall maximum sound level |
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398 and prints out that level when MPlayer exits. This volume estimate can be used |
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399 for setting the sound level in MEncoder such that the maximum dynamic range is |
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400 utilized.</P> |
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401 |
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402 |
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403 <H5><A NAME="af_equalizer">2.3.2.3.6 Equalizer</A></H5> |
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404 |
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405 <P> This filter is a 10 octave band graphic equalizer, implemented using 10 IIR |
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406 band pass filters. This means that it works regardless of what type of audio |
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407 is being played back. The center frequencies for the 10 bands are:</P> |
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408 |
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409 <TABLE BORDER="0" WIDTH="100%"> |
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410 <TR><TD>Band No.</TD><TD>Center frequency</TD></TR> |
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411 <TR><TD>0</TD><TD>31.25 Hz</TD></TR> |
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412 <TR><TD>1</TD><TD>62.50 Hz</TD></TR> |
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413 <TR><TD>2</TD><TD>125.0 Hz</TD></TR> |
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414 <TR><TD>3</TD><TD>250.0 Hz</TD></TR> |
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415 <TR><TD>4</TD><TD>500.0 Hz</TD></TR> |
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416 <TR><TD>5</TD><TD>1.000 kHz</TD></TR> |
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417 <TR><TD>6</TD><TD>2.000 kHz</TD></TR> |
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418 <TR><TD>7</TD><TD>4.000 kHz</TD></TR> |
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419 <TR><TD>8</TD><TD>8.000 kHz</TD></TR> |
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420 <TR><TD>9</TD><TD>16.00 kHz</TD></TR> |
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421 </TABLE> |
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422 |
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423 <P>If the sample rate of the sound being played back is lower than the center |
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424 frequency for a frequency band, then that band will be disabled. A known bug |
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425 with this filter is that the characteristics for the uppermost band are not |
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426 completely symmetric if the sample rate is close to the center frequency of |
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427 that band. This problem can be worked around by up-sampling the sound using |
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428 the resample filter before it reaches this filter. </P> |
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429 |
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430 <P> This filter has 10 parameters:</P> |
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431 |
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432 <DL> |
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433 <DT><CODE>g1:g2:g3...g10</CODE></DT> |
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434 <DD>are floating point numbers between -12 to +12dB representing the gain in |
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435 dB for each frequency band.</DD> |
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436 </DL> |
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437 |
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438 <P>Example:<BR> |
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439 <CODE>mplayer -af equalizer=11:11:10:5:0:-12:0:5:12:12 media.avi</CODE></P> |
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440 |
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441 <P>would amplify the sound in the upper and lower frequency region while |
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442 canceling it almost completely around 1kHz.</P> |
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443 |
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444 <H5><A NAME="af_panning">2.3.2.3.7 Panning filter </A></H5> |
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445 |
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446 <P>This filter can be used for mixing the channels arbitrarily. It is basically |
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447 a combination of the volume control and the channels filter. There are two |
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448 major uses for this filter: </P> |
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449 |
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450 <OL> |
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451 <LI>Down-mixing many channels to only a few, stereo to mono for example.</LI> |
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452 <LI>Varying the "width" of the center speaker in a surround sound system.</LI> |
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453 </OL> |
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454 |
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455 <P>This filter is hard to use, and will require some tinkering before the |
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456 desired result is obtained. The number of switches for this filter depends on |
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457 the number of output channels:</P> |
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458 |
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459 <DL> |
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460 <DT><CODE>nch</CODE></DT> |
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461 <DD>is an integer between 1 and 6 and is used for setting the number of output |
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462 channels. This switch is required, leaving it empty results in a runtime |
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463 error.</DD> |
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464 |
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465 <DT><CODE>l00:l01:l02:..l10:l11:l12:...ln0:ln1:ln2:...</CODE></DT> |
8757 | 466 <DD>are floating point values between 0 and 1. <CODE>l[i][j]</CODE> determines |
467 how much of input channel j is mixed into output channel i.</DD> | |
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468 </DL> |
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469 |
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470 <P>Example:<BR> |
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471 <CODE>mplayer -af pan=1:0.5:0.5 -channels 1 media.avi</CODE></P> |
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472 |
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473 <P>would down-mix from stereo to mono.</P> |
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474 |
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475 |
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476 <H2><STRONG>Note: Audio plugins have been deprecated by audio filters and will be |
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477 removed soon.</STRONG></H2> |
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478 |
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479 <H4><A NAME="plugins">2.3.2.4 Audio plugins (deprecated)</A></H4> |
3671 | 480 |
7814 | 481 <P>MPlayer has support for audio plugins. Audio plugins can be used for |
4508 | 482 changing the properties of the audio data before the sound reaches the sound |
5765 | 483 card. They are enabled using the <CODE>-aop</CODE> switch which takes a |
484 <CODE>list=plugin1,plugin2,...</CODE> argument. The <CODE>list</CODE> argument | |
485 is required and determines which plugins should be used and in which order they | |
6974 | 486 should be executed. Example:</P> |
4508 | 487 |
488 <P> <CODE>mplayer media.avi -aop list=resample,format</CODE></P> | |
489 | |
490 <P>would run the sound through the resampling plugin followed by the format | |
6974 | 491 plugin.</P> |
4508 | 492 |
5765 | 493 <P>The plugins can also have switches that change their behavior. These |
4508 | 494 switches are explained in detail in the sections below. A plugin will execute |
5765 | 495 using default settings if its switches are omitted. Here is an example of how |
6974 | 496 to use plugins in combination with plugin specific switches:</P> |
4508 | 497 |
498 <P> <CODE>mplayer media.avi -aop | |
6974 | 499 list=resample,format:fout=44100:format=0x8</CODE></P> |
4508 | 500 |
501 <P>would set the output frequency of the resample plugin to 44100Hz and the | |
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502 output format of the format plugin to AFMT_U8.</P> |
4508 | 503 |
7814 | 504 <P>Currently audio plugins can not be used in MEncoder.</P> |
4508 | 505 |
506 | |
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507 <H5><A NAME="resample">2.3.2.4.1 Up/Downsampling</A></H5> |
3671 | 508 |
7814 | 509 <P>MPlayer fully supports up/downsampling of the sound. This plugin can |
5765 | 510 be used if you have a fixed frequency sound card or if you are |
511 stuck with an old sound card that is only capable of max 44.1kHz. | |
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512 Whether is usage of this plugin is necessary or not, is <B>autodetected</B>. |
8260 | 513 This plugin has one switch: |
4508 | 514 <CODE>fout</CODE> which is used for setting the desired output sample |
5765 | 515 frequency. It defaults to 48kHz, and is given in |
6974 | 516 <Hz>.</P> |
3671 | 517 |
5765 | 518 <P>Usage:<BR> |
6974 | 519 <CODE>mplayer media.avi -aop list=resample:fout=<required |
3671 | 520 frequency in Hz, like 44100></CODE></P> |
521 | |
5765 | 522 <P>Note that the output frequency should not be scaled up from the default value. |
523 Scaling up will cause the audio and video streams to be played in slow motion | |
524 in addition to audio distortion.</P> | |
3671 | 525 |
6974 | 526 |
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527 <H5><A NAME="surround_decoding">2.3.2.4.2 Surround Sound decoding</A></H5> |
3931 | 528 |
7814 | 529 <P>MPlayer has an audio plugin that can decode matrix encoded |
6974 | 530 surround sound. Dolby Surround is an example of a matrix encoded format. |
531 Many files with 2 channel audio actually contain matrixed surround sound. | |
532 To use this feature you need a sound card supporting at least 4 channels.</P> | |
3931 | 533 |
5765 | 534 <P>Usage:<BR> |
6974 | 535 <CODE>mplayer media.avi -aop list=surround</CODE></P> |
3931 | 536 |
4508 | 537 |
8177 | 538 <H5><A NAME="format">2.3.2.3.3 Sample format converter</A></H5> |
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539 |
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540 <P>If your sound card driver does not support signed 16bit <CODE>int</CODE> data type, |
5765 | 541 this plugin can |
4508 | 542 be used to change the format to one which your sound card can understand. It |
5765 | 543 has one switch, <CODE>format</CODE>, which can be set to one of the numbers |
544 found in <CODE>libao2/afmt.h</CODE>. This plugin is hardly ever needed and is | |
545 intended for advanced users. Keep in mind that this plugin only changes the | |
6974 | 546 sample format and not the sample frequency or the number of channels.</P> |
4508 | 547 |
5765 | 548 <P>Usage:<BR> |
4508 | 549 <CODE>mplayer media.avi -aop |
6974 | 550 list=format:format=<required output format></CODE></P> |
4508 | 551 |
552 | |
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553 <H5><A NAME="delay">2.3.2.4.4 Delay</A></H5> |
4508 | 554 |
555 <P>This plugin delays the sound and is intended as an example of how to develop | |
5765 | 556 new plugins. It can not be used for anything useful from a users perspective |
4508 | 557 and is mentioned here for the sake of completeness only. Do not use this |
558 plugin unless you are a developer.</P> | |
559 | |
7047 | 560 |
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561 <H5><A NAME="volume">2.3.2.4.5 Software volume control</A></H5> |
4913 | 562 |
563 <P>This plugin is a software replacement for the volume control, and | |
5765 | 564 can be used on machines with a broken mixer device. It can also be |
7814 | 565 used if one wants to change the output volume of MPlayer |
4913 | 566 without changing the PCM volume setting in the mixer. It has one |
567 switch <CODE>volume</CODE> that is used for setting the initial | |
568 sound level. The initial sound level can be set to values between 0 | |
5099 | 569 and 255 and defaults to 101 which equals 0dB amplification. Use this |
570 plugin with caution since it can reduce the signal to noise ratio of | |
571 the sound. In most cases it is best to set the level for the PCM | |
572 sound to max, leave this plugin out and control the output level to | |
5765 | 573 your speakers with the master volume control of the mixer. If there is an |
5099 | 574 external amplifier connected to the computer (this is almost always |
575 the case), the noise level can be minimized by adjusting the master | |
576 level and the volume knob on the amplifier until the hissing noise | |
6974 | 577 in the background is gone.</P> |
4913 | 578 |
5765 | 579 <P>Usage:<BR> |
4913 | 580 <CODE>mplayer media.avi -aop |
6974 | 581 list=volume:volume=<0-255></CODE></P> |
4913 | 582 |
5765 | 583 <P>This plugin also has compressor or "soft-clipping" capabilities. |
5099 | 584 Compression can be used if the dynamic range of the sound is very |
585 high or if the dynamic range of the loudspeakers is very | |
5765 | 586 low. Be aware that this feature creates distortion and should be |
6974 | 587 considered a last resort.</P> |
5099 | 588 |
5765 | 589 <P>Usage:<BR> |
5099 | 590 <CODE>mplayer media.avi -aop |
6974 | 591 list=volume:softclip</CODE></P> |
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592 |
5109 | 593 |
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594 <H5><A NAME="extrastereo">2.3.2.4.6 Extrastereo</A></H5> |
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595 |
5765 | 596 <P>This plugin (linearly) increases the difference between left and right |
597 channels (like the XMMS extrastereo plugin) which gives some sort of "live" | |
6974 | 598 effect to playback.</P> |
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599 |
5765 | 600 <P>Usage:<BR> |
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601 <CODE>mplayer media.avi -aop list=extrastereo</CODE><BR> |
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602 <CODE>mplayer media.avi -aop list=extrastereo:mul=3.45</CODE></P> |
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603 |
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604 <P>The default coefficient (<CODE>mul</CODE>) is a float number that defaults |
5765 | 605 to 2.5. If you set it to 0.0, you will have mono sound (average of both |
6272 | 606 channels). If you set it to 1.0, sound will be unchanged, if you set it to |
607 -1.0, left and right channels will be swapped.</P> | |
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608 |
5109 | 609 |
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610 <H5><A NAME="normalizer">2.3.2.4.7 Volume normalizer</A></H5> |
5109 | 611 |
5765 | 612 <P>This plugin maximizes the volume without distorting the sound.</P> |
5109 | 613 |
5765 | 614 <P>Usage:<BR> |
5109 | 615 <CODE>mplayer media.avi -aop list=volnorm</CODE><BR> |
616 | |
5721 | 617 |
1612 | 618 </BODY> |
619 </HTML> |