Mercurial > mplayer.hg
annotate DOCS/sound.html @ 8750:f6323ff433aa
New audio filter documentation by Anders Johannsson with some structural
modifications by myself.
author | diego |
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date | Fri, 03 Jan 2003 22:29:16 +0000 |
parents | 352b654c7787 |
children | f117a4150786 |
rev | line source |
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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 |
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20 <TABLE BORDER="0"> |
6974 | 21 <TR><TD COLSPAN=4><P><B>General:</B></P></TD></TR> |
22 <TR><TD> </TD><TD VALIGN=top>oss</TD><TD> </TD><TD>OSS (ioctl) driver (supports hardware AC3 passthrough)</TD></TR> | |
8260 | 23 <TR><TD></TD><TD VALIGN=top>sdl</TD><TD></TD><TD>SDL driver (supports <B>ESD</B>, <B>ARTS</B> etc)</TD></TR> |
6974 | 24 <TR><TD></TD><TD VALIGN=top>nas</TD><TD></TD><TD>NAS (Network Audio System) driver</TD></TR> |
25 <TR><TD></TD><TD VALIGN=top>alsa5</TD><TD></TD><TD>native ALSA 0.5 driver</TD></TR> | |
26 <TR><TD></TD><TD VALIGN=top>alsa9</TD><TD></TD><TD>native ALSA 0.9 driver (supports hardware AC3 passthrough)</TD></TR> | |
27 <TR><TD></TD><TD VALIGN=top>sun</TD><TD></TD><TD>SUN audio driver (<CODE>/dev/audio</CODE>) for BSD and Solaris8 users</TD></TR> | |
28 <TR><TD></TD><TD VALIGN=top>arts</TD><TD></TD><TD>native ARTS driver (mostly for KDE users)</TD></TR> | |
1905 | 29 </TABLE> |
1612 | 30 |
8177 | 31 <P>Fact is, Linux sound card drivers have compatibility problems. The cause |
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32 is that MPlayer uses a feature that well coded audio drivers implement to |
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33 maintain audio/video sync. Regrettably, some driver authors do not care about |
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34 this function, it is not needed for playing MP3s or for sound effects.</P> |
8177 | 35 |
36 <P>Other media players like aviplay or xine possibly work out-of-the-box with | |
37 these drivers because they use "simple" methods with internal timing. A note: | |
38 time showed their methods aren't AS efficient as MPlayer's.</P> | |
39 | |
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40 <P>With a correctly written audio driver MPlayer will never create audio related |
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41 A/V desynchronisation, unless your file is badly broken. Some options to work |
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42 around these problems are described in the man page).</P> |
8177 | 43 |
44 <P>If you happen to have a bad audio driver, try the <CODE>-autosync</CODE> | |
45 option, it should sort out your problems. See the man page for detailed | |
46 information.</P> | |
47 | |
48 <P>Some notes:</P> | |
5765 | 49 |
50 <UL> | |
6960 | 51 <LI>If you have an OSS driver, first try <CODE>-ao oss</CODE> (this is the |
52 default). If you experience glitches, halts or anything out of the | |
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53 ordinary, try <CODE>-ao sdl</CODE> (NOTE: You need to have SDL libraries |
6960 | 54 and header files installed). The SDL audio driver helps in a lot of cases |
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55 and also supports ESD and ARTS. (ESD is the sound daemon |
6960 | 56 from GNOME, ARTS is from KDE.)</LI> |
57 <LI>If you have ALSA version 0.5, then you almost always have to use | |
58 <CODE>-ao alsa5</CODE> , since ALSA 0.5 has buggy OSS emulation code, and | |
59 will <B>crash MPlayer</B> with a message like this:<BR> | |
60 <CODE>DEMUXER: Too many (945 in 8390980 bytes) video packets in the buffer!</CODE></LI> | |
5765 | 61 </UL> |
1892 | 62 |
8177 | 63 <P>On <B>Solaris</B>, use the SUN audio driver with the |
6960 | 64 <CODE>-ao sun</CODE> option, otherwise neither video nor audio will work.</P> |
1892 | 65 |
7047 | 66 |
8177 | 67 <H4><A NAME="experiences">2.3.2.2 Sound Card experiences, recommendations</A></H4> |
1612 | 68 |
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69 <TABLE BORDER="0" WIDTH="100%"> |
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70 <TR><TD COLSPAN=3><B>VIA onboard chipset (via82cxxx) 48kHz only</B></TD></TR> |
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71 <TR><TD></TD><TD>Driver:</TD><TD> from the |
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72 <A HREF="http://sourceforge.net/project/showfiles.php?group_id=3242&release_id=59602">gkernel project</A></TD></TR> |
3190 | 73 |
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74 <TR><TD COLSPAN=3><B>Aureal Vortex 2</B></TD></TR> |
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75 <TR><TD> </TD><TD>OSS:</TD><TD>no driver</TD></TR> |
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76 <TR><TD></TD><TD>OSS/Pro:</TD><TD>OK</TD></TR> |
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77 <TR><TD></TD><TD>ALSA:</TD><TD>no driver</TD></TR> |
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78 <TR><TD></TD><TD>Max kHz:</TD><TD>48</TD></TR> |
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79 <TR><TD></TD><TD>Driver:</TD><TD><A HREF="http://aureal.sourceforge.net">aureal.sourceforge.net</A></TD></TR> |
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80 <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><BR> |
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81 (<I>buffer size increased to 32k</I>)</TD></TR> |
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83 <TR><TD COLSPAN=3><B>GUS PnP</B></TD></TR> |
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84 <TR><TD></TD><TD>OSS:</TD><TD>no driver</TD></TR> |
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85 <TR><TD></TD><TD>OSS/Pro:</TD><TD>OK</TD></TR> |
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86 <TR><TD></TD><TD>ALSA:</TD><TD>OK</TD></TR> |
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87 <TR><TD></TD><TD>Max kHz:</TD><TD>48</TD></TR> |
1682 | 88 |
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89 <TR><TD COLSPAN=3><B>SB Live!</B></TD></TR> |
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90 <TR><TD></TD><TD>OSS:</TD><TD>Analog OK, SP/DIF not working</TD></TR> |
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91 <TR><TD></TD><TD>ALSA:</TD><TD>Both OK</TD></TR> |
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92 <TR><TD></TD><TD>Max kHz:</TD><TD>192</TD></TR> |
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94 <TR><TD COLSPAN=3><B>SB AWE 64</B></TD></TR> |
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95 <TR><TD></TD><TD>OSS:</TD><TD>max 44kHz</TD></TR> |
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96 <TR><TD></TD><TD>ALSA:</TD><TD>48kHz sounds bad</TD></TR> |
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97 <TR><TD></TD><TD>Max kHz:</TD><TD>48</TD></TR> |
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99 <TR><TD COLSPAN=3><B>Gravis UltraSound ACE</B></TD></TR> |
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100 <TR><TD></TD><TD>OSS:</TD><TD>not 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>44</TD></TR> |
1612 | 103 |
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104 <TR><TD COLSPAN=3><B>Gravis UltraSound MAX</B></TD></TR> |
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105 <TR><TD></TD><TD>OSS:</TD><TD>OK</TD></TR> |
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106 <TR><TD></TD><TD>ALSA:</TD><TD>OK (?)</TD></TR> |
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107 <TR><TD></TD><TD>Max kHz:</TD><TD>48</TD></TR> |
1612 | 108 |
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109 <TR><TD COLSPAN=3><B>ESS 688</B></TD></TR> |
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110 <TR><TD></TD><TD>OSS:</TD><TD>OK</TD></TR> |
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111 <TR><TD></TD><TD>ALSA:</TD><TD>OK (?)</TD></TR> |
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112 <TR><TD></TD><TD>Max kHz:</TD><TD>48</TD></TR> |
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114 <TR><TD COLSPAN=3><B>C-Media cards (which ones?)</B></TD></TR> |
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115 <TR><TD></TD><TD>OSS:</TD><TD>not OK (hissing) (?)</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>?</TD></TR> |
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119 <TR><TD COLSPAN=3><B>Yamaha cards (*ymf*)</B></TD></TR> |
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120 <TR><TD></TD><TD>OSS:</TD><TD>not OK (?) (maybe <CODE>-ao sdl</CODE>)</TD></TR> |
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121 <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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122 <TR><TD></TD><TD>Max kHz:</TD><TD>?</TD></TR> |
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124 <TR><TD COLSPAN=3><B>Cards with envy24 chips (like Terratec EWS88MT)</B></TD></TR> |
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125 <TR><TD></TD><TD>OSS:</TD><TD>?</TD></TR> |
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126 <TR><TD></TD><TD>OSS/Pro:</TD><TD>OK</TD></TR> |
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127 <TR><TD></TD><TD>ALSA:</TD><TD>?</TD></TR> |
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128 <TR><TD></TD><TD>Max kHz:</TD><TD>?</TD></TR> |
3438 | 129 |
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130 <TR><TD COLSPAN=3><B>PC Speaker or DAC</B></TD></TR> |
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131 <TR><TD></TD><TD>OSS:</TD><TD>OK (Use the SDL driver: <CODE>-ao sdl</CODE>)</TD></TR> |
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132 <TR><TD></TD><TD>ALSA:</TD><TD>no driver</TD></TR> |
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133 <TR><TD></TD><TD>Max kHz:</TD><TD>The driver emulates 44.1, maybe more.</TD></TR> |
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134 <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 | 135 </TABLE> |
1612 | 136 |
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137 <P>On Linux, a 2.4.x kernel is highly recommended. Kernel 2.2 is not tested.</P> |
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138 |
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139 <P>If the sound clicks when playing from CD-ROM, turn on IRQ unmasking, e.g. |
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140 <CODE>hdparm -u1 /dev/cdrom</CODE> (<CODE>man hdparm</CODE>). This is |
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141 generally beneficial and described in more detail in the |
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142 <A HREF="cd-dvd.html#drives">CD-ROM section</A>.</P> |
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143 |
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144 <P>Sharing your sound card with another application like XMMS is <B>strongly |
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145 discouraged</B>! If the other sound application is using ESD, start |
7814 | 146 MPlayer with the <CODE>-vo sdl:esd</CODE> option to combine both |
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147 sound streams. In fact, the option <CODE>-vo sdl:esd</CODE> could be used |
7814 | 148 with ESD even when playing MPlayer alone.</P> |
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149 |
7814 | 150 <P>Feedback to this document is welcome. Please tell us how MPlayer |
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151 and your sound card(s) worked together.</P> |
1612 | 152 |
3671 | 153 |
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154 <H4><A NAME="af">2.3.2.3 Audio filters</A></H4> |
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155 |
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156 <P>The old audio plugins have been superseded by a new audio filter layer. Audio |
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157 filters are used for changing the properties of the audio data before the |
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158 sound reaches the sound card. The activation and deactivation of the filters |
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159 is normally automated but can be overridden. The filters are activated when |
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160 the properties of the audio data differ from those required by the sound card |
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161 and deactivated if unnecessary. The <CODE>-af filter1,filter2,...</CODE> |
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162 switch is used to override the automatic activation of filters or to insert |
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163 filters that are not automatically inserted. The filters will be executed as |
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164 they appear in the comma separated list.</P> |
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165 |
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166 <P>Example:<BR> |
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167 <CODE>mplayer -af resample,pan movie.avi </CODE></P> |
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168 |
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169 <P>would run the sound through the resampling filter followed by the pan filter. |
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170 Observe that the list must not contain any spaces, else it will fail.</P> |
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171 |
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172 <P>The filters often have switches that change their behavior. These switches |
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173 are explained in detail in the sections below. A filter will execute using |
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174 default settings if its switches are omitted. Here is an example of how to use |
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175 filters in combination with filter specific switches:</P> |
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176 |
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177 <P> <CODE>mplayer -af resample=11025,pan=1:0.5:0.5 -channels 1 |
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178 -srate 11025 media.avi</CODE></P> |
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179 |
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180 <P>would set the output frequency of the resample filter to 11025Hz and downmix |
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181 the audio to 1 channel using the pan filter.</P> |
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182 |
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183 <P>Most filters respond to the <CODE>-v</CODE> switch, which makes the filters |
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184 print out status messages.</P> |
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185 |
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186 <P>The overall execution of the filter layer is controlled using the |
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187 <CODE>-af-adv</CODE> switch. This switch has two suboptions:</P> |
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188 |
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189 <DL> |
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190 <DT><CODE>force</CODE><DT> |
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191 <DD>is an integer between 0 and 3 that controls how the filters are inserted |
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192 and what speed/accuracy optimizations they use: |
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193 <DL> |
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194 <DT>0</DT> |
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195 <DD>Use automatic insertion of filters and optimize according to CPU |
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196 speed.</DD> |
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197 <DT>1</DT> |
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198 <DD>Use automatic insertion of filters and optimize for the highest speed. |
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199 If this option is set the processing of the audio data will be done |
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200 using fix point arithmetics. Warning: Some features in the audio filters |
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201 will silently fail, and the sound quality may drop.</DD> |
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202 <DT>2</DT> |
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203 <DD>Use automatic insertion of filters and optimize for quality. If this |
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204 option is set the processing of the audio data will be done using |
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205 floating point instructions and is therefore quite CPU intensive, but |
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206 gives a lot higher sound quality than fix point processing.</DD> |
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207 <DT>3</DT> |
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208 <DD>Use no automatic insertion of filters and no optimization. Warning: It |
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209 may be possible to crash MPlayer using this setting.</DD> |
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210 </DL> |
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211 |
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212 </DD> |
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213 |
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214 <DT><CODE>list</CODE></DT> |
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215 <DD>is an alias for the -af switch.</DD> |
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216 </DL> |
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217 |
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218 |
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219 <H5><A NAME="af_resample">2.3.2.3.1 Up/Down-sampling</A></H5> |
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220 |
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221 <P>MPlayer fully supports sound up/down-sampling. This filter can be used if you |
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222 have a fixed frequency sound card or if you are stuck with an old sound card |
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223 that is only capable of max 44.1kHz. This filter is automatically enabled if |
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224 it is necessary, but it can also be explicitly enabled on the command line. It |
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225 has three switches:</P> |
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226 |
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227 <DL> |
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228 <DT><CODE>srate</CODE></DT> |
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229 <DD>is an integer used for setting the output sample |
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230 frequency in Hz. The valid range for this parameter is 8kHz to 192kHz. If |
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231 the input and output sample frequency are the same or if this parameter is |
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232 omitted the filter is automatically unloaded. A high sample frequency |
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233 normally improves the audio quality, especially when used in combination |
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234 with other filters.</DD> |
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235 |
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236 <DT><CODE>sloppy</CODE></DT> |
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237 <DD>is an optional binary parameter that allows the output frequency to differ |
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238 slightly from the frequency given by <CODE>srate</CODE>. This switch can be |
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239 used if the startup of the playback is extremely slow.</DD> |
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240 |
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241 <DT><CODE>fast</CODE><DT> |
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242 <DD>is an optional binary parameter that enables linear interpolation as |
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243 resampling method. Linear interpolation is extremely fast, but suffers from |
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244 poor sound quality especially when used for up-sampling.</DD> |
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245 </DL> |
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246 |
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247 <P>Example:<BR> |
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248 <CODE>mplayer -af resample=44100:0:1</CODE></P> |
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249 |
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250 <P>would set the output frequency of the resample filter to 44100Hz using exact |
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251 output frequency scaling and linear interpolation.</P> |
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252 |
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253 |
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254 <H5><A NAME="af_channels">2.3.2.3.2 Changing the number of channels</A></H5> |
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255 |
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256 <P>The <CODE>channels</CODE> filter can be used for adding and removing |
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257 channels, it can also be used for routing or copying channels. It is |
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258 automatically enabled when the output from the audio filter layer differs from |
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259 the input layer or when it is requested by another filter. This filter unloads |
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260 itself if not needed. The number of switches is dynamic:</P> |
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261 |
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262 <DL> |
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263 <DT><CODE>nch</CODE></DT> |
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264 <DD>is an integer between 1 and 6 that is used for setting the number of |
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265 output channels. This switch is required, leaving it empty results in a |
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266 runtime error.</DD> |
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267 |
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268 <DT><CODE>nr</CODE></DT> |
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269 <DD>is an integer between 1 and 6 that is used for specifying the number of |
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270 routes. This parameter is optional. If it is omitted the default routing is |
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271 used.</DD> |
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272 |
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273 <DT><CODE>from1:to1:from2:to2:from3:to3...</CODE></DT> |
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274 <DD>are pairs of numbers between 0 and 5 that define where each channel should |
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275 be routed.</DD> |
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276 </DL> |
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277 |
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278 <P>If only <CODE>nch</CODE> is given the default routing is used, it works as |
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279 follows: If the number of output channels is bigger than the number of input |
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280 channels empty channels are inserted (except mixing from mono to stereo, then |
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281 the mono channel is repeated in both of the output channels). If the number of |
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282 output channels is smaller than the number of input channels the exceeding |
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283 channels are truncated.</P> |
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284 |
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285 <P>Example 1:<BR> |
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286 <CODE>mplayer -af channels=4:4:0:1:1:0:2:2:3:3 media.avi </CODE></P> |
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287 |
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288 <P>would change the number of channels to 4 and set up 4 routes that swap |
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289 channel 0 and channel 1 and leave channel 2 and 3 intact. Observe that if |
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290 media containing two channels was played back, channels 2 and 3 would contain |
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291 silence but 0 and 1 would still be swapped.</P> |
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292 |
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293 <P>Example 2:<BR> |
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294 <CODE>mplayer -af channels=6:4:0:0:0:1:0:2:0:3 media.avi </CODE></P> |
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295 |
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296 <P>would change the number of channels to 6 and set up 4 routes that copy |
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297 channel 0 to channels 0 to 3. Channel 4 and 5 will contain silence.</P> |
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298 |
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299 |
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300 <H5><A NAME="af_format">2.3.2.3.3 Sample format converter</A></H5> |
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301 |
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302 <P>This filter is a sample format converter. It is automatically enabled when |
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303 needed by the sound card or another filter.</P> |
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304 |
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305 <DL> |
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306 <DT><CODE>bps</CODE></DT> |
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307 <DD>can be 1, 2 or 4 and denotes the number of bytes per sample. This switch |
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308 is required, leaving it empty results in a runtime error.</DD> |
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309 |
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310 <DT><CODE>f</CODE></DT> |
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311 <DD>is a text string describing the sample format. The string is a |
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312 concatenated mix of: <CODE>alaw</CODE>, <CODE>mulaw</CODE> or |
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313 <CODE>imaadpcm</CODE>, <CODE>float</CODE> or <CODE>int</CODE>, |
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314 <CODE>unsigned</CODE> or <CODE>signed</CODE>, <CODE>le</CODE> or |
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315 <CODE>be</CODE> (little or big endian). This switch is required, leaving it |
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316 empty results in a runtime error.</DD> |
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317 </DL> |
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318 |
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319 <P>Example:<BR> |
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320 <CODE>mplayer media.avi -af format=4:float</CODE></P> |
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321 |
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322 <P>would set the output output format to 4 bytes per sample floating point |
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323 data.</P> |
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324 |
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325 |
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326 <H5><A NAME="af_delay">2.3.2.3.4 Delay</A></H5> |
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327 |
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328 <P>This filter delays the sound to the loudspeakers in order to make the sound |
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329 in the different channels arrive at the same time to the listening position. |
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330 It is only useful if you have more than 2 loudspeakers. This filter has a |
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331 variable number of parameters:</P> |
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332 |
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333 <DL> |
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334 <DT><CODE>d1:d2:d3...</CODE></DT> |
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335 <DD>are floating point numbers representing the delays in ms that should be |
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336 imposed on the different channels. The minimum delay is 0ms and the maximum |
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337 is 1000ms.</DD> |
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338 </DL> |
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339 |
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340 <P>To calculate the required delay for the different channels do as follows:</P> |
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341 |
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342 <OL> |
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343 <LI>Measure the distance to the loudspeakers in meters in relation to your |
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344 listening position, giving you the distances s1 to s5 (for a 5.1 system). |
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345 There is no point in compensating for the sub-woofer (you will not hear the |
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346 difference anyway).</LI> |
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347 <LI>Subtract the distances s1 to s5 from the maximum distance i.e.<BR> |
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348 s[i] = max(s) - s[i]; i = 1...5</LI> |
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349 <LI>Calculated the required delays in ms as<BR> |
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350 d[i] = 1000*s[i]/342; i = 1...5 </LI> |
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351 </OL> |
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352 |
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353 <P>Example:<BR> |
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354 <CODE>mplayer -af delay=10.5:10.5:0:0:7:0 media.avi</CODE></P> |
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355 |
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356 <P>would delay front left and right by 10.5ms, the two rear channels and the sub |
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357 by 0ms and the center channel by 7ms.</P> |
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358 |
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359 |
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360 <H5><A NAME="af_volume">2.3.2.3.5 Software volume control</A></H5> |
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361 |
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362 <P>This filter is a software volume control. Use this filter with caution since |
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363 it can reduce the signal to noise ratio of the sound. In most cases it is best |
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364 to set the level for the PCM sound to max, leave this filter out and control |
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365 the output level to your speakers with the master volume control of the mixer. |
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366 If there is an external amplifier connected to the computer (this is almost |
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367 always the case), the noise level can be minimized by adjusting the master |
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368 level and the volume knob on the amplifier until the hissing noise in the |
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369 background is gone. This filter has two switches:</P> |
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370 |
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371 <DL> |
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372 <DT><CODE>v</CODE></DT> |
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373 <DD>is a floating point number between -200 and +60 which represents the |
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374 volume level in dB. The default level is -10dB.</DD> |
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375 |
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376 <DT><CODE>c</CODE></DT> |
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377 <DD>is a binary control that turns soft clipping on and off. Soft-clipping can |
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378 make the sound more smooth if very high volume levels are used. Enable this |
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379 switch if the dynamic range of the loudspeakers is very low. Be aware that |
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380 this feature creates distortion and should be considered a last resort.</DD> |
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381 </DL> |
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382 |
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383 <P>Example:<BR> |
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384 <CODE>mplayer -af volume=10.1:0 media.avi</CODE></P> |
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385 |
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386 <P>would amplify the sound by 10.1dB and hard-clip if the sound level is too |
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387 high.</P> |
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388 |
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389 <P>This filter has a second feature: It measures the overall maximum sound level |
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390 and prints out that level when MPlayer exits. This volume estimate can be used |
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391 for setting the sound level in MEncoder such that the maximum dynamic range is |
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392 utilized.</P> |
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393 |
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394 |
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395 <H5><A NAME="af_equalizer">2.3.2.3.6 Equalizer</A></H5> |
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396 |
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397 <P> This filter is a 10 octave band graphic equalizer, implemented using 10 IIR |
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398 band pass filters. This means that it works regardless of what type of audio |
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399 is being played back. The center frequencies for the 10 bands are:</P> |
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400 |
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401 <TABLE BORDER="0" WIDTH="100%"> |
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402 <TR><TD>Band No.</TD><TD>Center frequency</TD></TR> |
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403 <TR><TD>0</TD><TD>31.25 Hz</TD></TR> |
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404 <TR><TD>1</TD><TD>62.50 Hz</TD></TR> |
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405 <TR><TD>2</TD><TD>125.0 Hz</TD></TR> |
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406 <TR><TD>3</TD><TD>250.0 Hz</TD></TR> |
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407 <TR><TD>4</TD><TD>500.0 Hz</TD></TR> |
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408 <TR><TD>5</TD><TD>1.000 kHz</TD></TR> |
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409 <TR><TD>6</TD><TD>2.000 kHz</TD></TR> |
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410 <TR><TD>7</TD><TD>4.000 kHz</TD></TR> |
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411 <TR><TD>8</TD><TD>8.000 kHz</TD></TR> |
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412 <TR><TD>9</TD><TD>16.00 kHz</TD></TR> |
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413 </TABLE> |
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414 |
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415 <P>If the sample rate of the sound being played back is lower than the center |
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416 frequency for a frequency band, then that band will be disabled. A known bug |
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417 with this filter is that the characteristics for the uppermost band are not |
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418 completely symmetric if the sample rate is close to the center frequency of |
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419 that band. This problem can be worked around by up-sampling the sound using |
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420 the resample filter before it reaches this filter. </P> |
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421 |
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422 <P> This filter has 10 parameters:</P> |
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423 |
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424 <DL> |
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425 <DT><CODE>g1:g2:g3...g10</CODE></DT> |
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426 <DD>are floating point numbers between -12 to +12dB representing the gain in |
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427 dB for each frequency band.</DD> |
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428 </DL> |
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429 |
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430 <P>Example:<BR> |
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431 <CODE>mplayer -af equalizer=11:11:10:5:0:-12:0:5:12:12 media.avi</CODE></P> |
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432 |
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433 <P>would amplify the sound in the upper and lower frequency region while |
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434 canceling it almost completely around 1kHz.</P> |
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435 |
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436 <H5><A NAME="af_panning">2.3.2.3.7 Panning filter </A></H5> |
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437 |
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438 <P>This filter can be used for mixing the channels arbitrarily. It is basically |
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439 a combination of the volume control and the channels filter. There are two |
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440 major uses for this filter: </P> |
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441 |
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442 <OL> |
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443 <LI>Down-mixing many channels to only a few, stereo to mono for example.</LI> |
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444 <LI>Varying the "width" of the center speaker in a surround sound system.</LI> |
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445 </OL> |
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446 |
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447 <P>This filter is hard to use, and will require some tinkering before the |
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448 desired result is obtained. The number of switches for this filter depends on |
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449 the number of output channels:</P> |
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450 |
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451 <DL> |
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452 <DT><CODE>nch</CODE></DT> |
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453 <DD>is an integer between 1 and 6 and is used for setting the number of output |
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454 channels. This switch is required, leaving it empty results in a runtime |
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455 error.</DD> |
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456 |
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457 <DT><CODE>l00:l01:l02:..l10:l11:l12:...ln0:ln1:ln2:...</CODE></DT> |
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458 <DD>are floating point values between 0 and 1 that determine the level |
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459 <CODE>l[i][j]</CODE> that the input channel j is mixed into output channel |
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460 i.</DD> |
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461 </DL> |
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462 |
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463 <P>Example:<BR> |
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464 <CODE>mplayer -af pan=1:0.5:0.5 -channels 1 media.avi</CODE></P> |
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465 |
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466 <P>would down-mix from stereo to mono.</P> |
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467 |
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468 |
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469 <H2><STRONG>Note: Audio plugins have been deprecated by audio filters and will be |
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470 removed soon.</STRONG></H2> |
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471 |
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472 <H4><A NAME="plugins">2.3.2.4 Audio plugins (deprecated)</A></H4> |
3671 | 473 |
7814 | 474 <P>MPlayer has support for audio plugins. Audio plugins can be used for |
4508 | 475 changing the properties of the audio data before the sound reaches the sound |
5765 | 476 card. They are enabled using the <CODE>-aop</CODE> switch which takes a |
477 <CODE>list=plugin1,plugin2,...</CODE> argument. The <CODE>list</CODE> argument | |
478 is required and determines which plugins should be used and in which order they | |
6974 | 479 should be executed. Example:</P> |
4508 | 480 |
481 <P> <CODE>mplayer media.avi -aop list=resample,format</CODE></P> | |
482 | |
483 <P>would run the sound through the resampling plugin followed by the format | |
6974 | 484 plugin.</P> |
4508 | 485 |
5765 | 486 <P>The plugins can also have switches that change their behavior. These |
4508 | 487 switches are explained in detail in the sections below. A plugin will execute |
5765 | 488 using default settings if its switches are omitted. Here is an example of how |
6974 | 489 to use plugins in combination with plugin specific switches:</P> |
4508 | 490 |
491 <P> <CODE>mplayer media.avi -aop | |
6974 | 492 list=resample,format:fout=44100:format=0x8</CODE></P> |
4508 | 493 |
494 <P>would set the output frequency of the resample plugin to 44100Hz and the | |
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495 output format of the format plugin to AFMT_U8.</P> |
4508 | 496 |
7814 | 497 <P>Currently audio plugins can not be used in MEncoder.</P> |
4508 | 498 |
499 | |
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500 <H5><A NAME="resample">2.3.2.4.1 Up/Downsampling</A></H5> |
3671 | 501 |
7814 | 502 <P>MPlayer fully supports up/downsampling of the sound. This plugin can |
5765 | 503 be used if you have a fixed frequency sound card or if you are |
504 stuck with an old sound card that is only capable of max 44.1kHz. | |
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505 Whether is usage of this plugin is necessary or not, is <B>autodetected</B>. |
8260 | 506 This plugin has one switch: |
4508 | 507 <CODE>fout</CODE> which is used for setting the desired output sample |
5765 | 508 frequency. It defaults to 48kHz, and is given in |
6974 | 509 <Hz>.</P> |
3671 | 510 |
5765 | 511 <P>Usage:<BR> |
6974 | 512 <CODE>mplayer media.avi -aop list=resample:fout=<required |
3671 | 513 frequency in Hz, like 44100></CODE></P> |
514 | |
5765 | 515 <P>Note that the output frequency should not be scaled up from the default value. |
516 Scaling up will cause the audio and video streams to be played in slow motion | |
517 in addition to audio distortion.</P> | |
3671 | 518 |
6974 | 519 |
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520 <H5><A NAME="surround_decoding">2.3.2.4.2 Surround Sound decoding</A></H5> |
3931 | 521 |
7814 | 522 <P>MPlayer has an audio plugin that can decode matrix encoded |
6974 | 523 surround sound. Dolby Surround is an example of a matrix encoded format. |
524 Many files with 2 channel audio actually contain matrixed surround sound. | |
525 To use this feature you need a sound card supporting at least 4 channels.</P> | |
3931 | 526 |
5765 | 527 <P>Usage:<BR> |
6974 | 528 <CODE>mplayer media.avi -aop list=surround</CODE></P> |
3931 | 529 |
4508 | 530 |
8177 | 531 <H5><A NAME="format">2.3.2.3.3 Sample format converter</A></H5> |
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532 |
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533 <P>If your sound card driver does not support signed 16bit <CODE>int</CODE> data type, |
5765 | 534 this plugin can |
4508 | 535 be used to change the format to one which your sound card can understand. It |
5765 | 536 has one switch, <CODE>format</CODE>, which can be set to one of the numbers |
537 found in <CODE>libao2/afmt.h</CODE>. This plugin is hardly ever needed and is | |
538 intended for advanced users. Keep in mind that this plugin only changes the | |
6974 | 539 sample format and not the sample frequency or the number of channels.</P> |
4508 | 540 |
5765 | 541 <P>Usage:<BR> |
4508 | 542 <CODE>mplayer media.avi -aop |
6974 | 543 list=format:format=<required output format></CODE></P> |
4508 | 544 |
545 | |
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546 <H5><A NAME="delay">2.3.2.4.4 Delay</A></H5> |
4508 | 547 |
548 <P>This plugin delays the sound and is intended as an example of how to develop | |
5765 | 549 new plugins. It can not be used for anything useful from a users perspective |
4508 | 550 and is mentioned here for the sake of completeness only. Do not use this |
551 plugin unless you are a developer.</P> | |
552 | |
7047 | 553 |
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554 <H5><A NAME="volume">2.3.2.4.5 Software volume control</A></H5> |
4913 | 555 |
556 <P>This plugin is a software replacement for the volume control, and | |
5765 | 557 can be used on machines with a broken mixer device. It can also be |
7814 | 558 used if one wants to change the output volume of MPlayer |
4913 | 559 without changing the PCM volume setting in the mixer. It has one |
560 switch <CODE>volume</CODE> that is used for setting the initial | |
561 sound level. The initial sound level can be set to values between 0 | |
5099 | 562 and 255 and defaults to 101 which equals 0dB amplification. Use this |
563 plugin with caution since it can reduce the signal to noise ratio of | |
564 the sound. In most cases it is best to set the level for the PCM | |
565 sound to max, leave this plugin out and control the output level to | |
5765 | 566 your speakers with the master volume control of the mixer. If there is an |
5099 | 567 external amplifier connected to the computer (this is almost always |
568 the case), the noise level can be minimized by adjusting the master | |
569 level and the volume knob on the amplifier until the hissing noise | |
6974 | 570 in the background is gone.</P> |
4913 | 571 |
5765 | 572 <P>Usage:<BR> |
4913 | 573 <CODE>mplayer media.avi -aop |
6974 | 574 list=volume:volume=<0-255></CODE></P> |
4913 | 575 |
5765 | 576 <P>This plugin also has compressor or "soft-clipping" capabilities. |
5099 | 577 Compression can be used if the dynamic range of the sound is very |
578 high or if the dynamic range of the loudspeakers is very | |
5765 | 579 low. Be aware that this feature creates distortion and should be |
6974 | 580 considered a last resort.</P> |
5099 | 581 |
5765 | 582 <P>Usage:<BR> |
5099 | 583 <CODE>mplayer media.avi -aop |
6974 | 584 list=volume:softclip</CODE></P> |
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585 |
5109 | 586 |
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587 <H5><A NAME="extrastereo">2.3.2.4.6 Extrastereo</A></H5> |
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588 |
5765 | 589 <P>This plugin (linearly) increases the difference between left and right |
590 channels (like the XMMS extrastereo plugin) which gives some sort of "live" | |
6974 | 591 effect to playback.</P> |
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592 |
5765 | 593 <P>Usage:<BR> |
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594 <CODE>mplayer media.avi -aop list=extrastereo</CODE><BR> |
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595 <CODE>mplayer media.avi -aop list=extrastereo:mul=3.45</CODE></P> |
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596 |
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597 <P>The default coefficient (<CODE>mul</CODE>) is a float number that defaults |
5765 | 598 to 2.5. If you set it to 0.0, you will have mono sound (average of both |
6272 | 599 channels). If you set it to 1.0, sound will be unchanged, if you set it to |
600 -1.0, left and right channels will be swapped.</P> | |
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601 |
5109 | 602 |
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603 <H5><A NAME="normalizer">2.3.2.4.7 Volume normalizer</A></H5> |
5109 | 604 |
5765 | 605 <P>This plugin maximizes the volume without distorting the sound.</P> |
5109 | 606 |
5765 | 607 <P>Usage:<BR> |
5109 | 608 <CODE>mplayer media.avi -aop list=volnorm</CODE><BR> |
609 | |
5721 | 610 |
1612 | 611 </BODY> |
612 </HTML> |