Mercurial > mplayer.hg
annotate DOCS/sound.html @ 8950:8fd72eae8886
Small updates for correctness and consistency.
author | diego |
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date | Mon, 13 Jan 2003 23:45:14 +0000 |
parents | 5b73c925436e |
children | ebc368360ea2 |
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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> |
1612 | 133 |
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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>The overall execution of the filter layer is controlled using the |
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193 <CODE>-af-adv</CODE> switch. This switch has two suboptions:</P> |
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194 |
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195 <DL> |
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196 <DT><CODE>force</CODE><DT> |
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197 <DD>is a Bit field that controls how the filters are inserted and what |
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198 speed/accuracy optimizations they use: |
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199 <DL> |
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200 <DT><CODE>0</CODE></DT> |
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201 <DD>Use automatic insertion of filters and optimize according to CPU |
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202 speed.</DD> |
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203 <DT><CODE>1</CODE></DT> |
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204 <DD>Use automatic insertion of filters and optimize for the highest |
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205 speed.<BR> |
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206 <EM>Warning:</EM> Some features in the audio filters may silently fail, |
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207 and the sound quality may drop.</DD> |
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208 <DT><CODE>2</CODE></DT> |
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209 <DD>Use automatic insertion of filters and optimize for quality.</DD> |
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210 <DT><CODE>3</CODE></DT> |
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211 <DD>Use no automatic insertion of filters and no optimization.<BR> |
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212 <I>Warning:</I> It may be possible to crash MPlayer using this |
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213 setting.</DD> |
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214 <DT><CODE>4</CODE></DT> |
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215 <DD>Use automatic insertion of filters according to 0 above, but use |
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216 floating point processing when possible.</DD> |
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217 <DT><CODE>5</CODE></DT> |
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218 <DD>Use automatic insertion of filters according to 1 above, but use |
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219 floating point processing when possible.</DD> |
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220 <DT><CODE>6</CODE></DT> |
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221 <DD>Use automatic insertion of filters according to 2 above, but use |
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222 floating point processing when possible.</DD> |
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223 <DT><CODE>7</CODE></DT> |
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224 <DD>Use no automatic insertion of filters according to 3 above, and use |
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225 floating point processing when possible.</DD> |
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226 </DL> |
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227 </DD> |
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228 |
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229 <DT><CODE>list</CODE></DT> |
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230 <DD>is an alias for the -af switch.</DD> |
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231 </DL> |
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232 |
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233 <P>The filter layer is also affected by the following generic switches: |
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234 |
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235 <DL> |
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236 <DT><CODE>-v</CODE></DT> |
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237 <DD>Increases the verbosity level and makes most filters print out extra |
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238 status messages.</DD> |
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239 <DT><CODE>-channels</CODE></DT> |
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240 <DD>This option sets the number of output channels your sound card is using. |
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241 It also affects the number of channels that are being decoded from the |
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242 media. If the media contains less channels than requested the channels |
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243 filter (see below) will automatically be inserted. The routing will be the |
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244 default routing for the channels filter.</DD> |
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245 <DT><CODE>-srate</CODE></DT> |
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246 <DD>This option selects the sample rate of your sound card. If the sample |
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247 frequency of your sound card is different from that of the current media, |
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248 the resample filter (see below) will be inserted into the audio filter layer |
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249 to compensate for the difference.</DD> |
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250 <DT><CODE>-format</CODE><DT> |
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251 <DD>This option sets the sample format of the audio filter layer and the sound |
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252 card. If the requested sample format of your sound card is different from |
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253 that of the current media, a format filter (see below) will be inserted to |
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254 rectify the difference.</DD> |
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255 </DL> |
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256 |
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257 |
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258 <H5><A NAME="af_resample">2.3.2.3.1 Up/Down-sampling</A></H5> |
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259 |
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261 <CODE>resample</CODE> filter. It can be used if you | |
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262 have a fixed frequency sound card or if you are stuck with an old sound card |
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263 that is only capable of max 44.1kHz. This filter is automatically enabled if |
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264 it is necessary, but it can also be explicitly enabled on the command line. It |
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265 has three switches:</P> |
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266 |
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267 <DL> |
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269 <DD>is an integer used for setting the output sample |
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270 frequency in Hz. The valid range for this parameter is 8kHz to 192kHz. If |
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271 the input and output sample frequency are the same or if this parameter is |
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272 omitted the filter is automatically unloaded. A high sample frequency |
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273 normally improves the audio quality, especially when used in combination |
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274 with other filters.</DD> |
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275 |
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276 <DT><CODE>sloppy</CODE></DT> |
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277 <DD>is an optional binary parameter that allows the output frequency to differ |
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278 slightly from the frequency given by <CODE>srate</CODE>. This switch can be |
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279 used if the startup of the playback is extremely slow. It is enabled by |
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280 default.</DD> |
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281 |
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282 <DT><CODE>type <0-2></CODE><DT> |
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283 <DD>is an optional integer between <CODE>0</CODE> and <CODE>2</CODE> that |
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284 selects which resampling method to use. Here <CODE>0</CODE> represents |
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285 linear interpolation as resampling method, <CODE>1</CODE> represents |
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286 resampling using a poly-phase filter-bank and integer processing and |
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287 <CODE>2</CODE> represents resampling using a poly-phase filter-bank and |
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288 floating point processing. Linear interpolation is extremely fast, but |
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289 suffers from poor sound quality especially when used for up-sampling. The |
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290 best quality is given by <CODE>2</CODE> but this method also suffers from |
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291 the highest CPU load.</DD> |
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292 </DL> |
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293 |
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294 <P>Example:<BR> |
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296 |
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297 <P>would set the output frequency of the resample filter to 44100Hz using exact |
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298 output frequency scaling and linear interpolation.</P> |
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299 |
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300 |
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301 <H5><A NAME="af_channels">2.3.2.3.2 Changing the number of channels</A></H5> |
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302 |
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303 <P>The <CODE>channels</CODE> filter can be used for adding and removing |
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304 channels, it can also be used for routing or copying channels. It is |
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305 automatically enabled when the output from the audio filter layer differs from |
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306 the input layer or when it is requested by another filter. This filter unloads |
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307 itself if not needed. The number of switches is dynamic:</P> |
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308 |
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309 <DL> |
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310 <DT><CODE>nch <1-6></CODE></DT> |
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311 <DD>is an integer between <CODE>1</CODE> and <CODE>6</CODE> that is used for |
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312 setting the number of output channels. This switch is required, leaving it |
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313 empty results in a runtime error.</DD> |
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314 |
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315 <DT><CODE>nr <1-6></CODE></DT> |
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316 <DD>is an integer between <CODE>1</CODE> and <CODE>6</CODE> that is used for |
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317 specifying the number of routes. This parameter is optional. If it is |
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318 omitted the default routing is used.</DD> |
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319 |
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320 <DT><CODE>from1:to1:from2:to2:from3:to3...</CODE></DT> |
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321 <DD>are pairs of numbers between <CODE>0</CODE> and <CODE>5</CODE> that define |
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322 where each channel should be routed.</DD> |
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323 </DL> |
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324 |
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325 <P>If only <CODE>nch</CODE> is given the default routing is used, it works as |
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326 follows: If the number of output channels is bigger than the number of input |
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327 channels empty channels are inserted (except mixing from mono to stereo, then |
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328 the mono channel is repeated in both of the output channels). If the number of |
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329 output channels is smaller than the number of input channels the exceeding |
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330 channels are truncated.</P> |
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331 |
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332 <P>Example 1:<BR> |
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333 <CODE>mplayer -af channels=4:4:0:1:1:0:2:2:3:3 media.avi </CODE></P> |
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334 |
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335 <P>would change the number of channels to 4 and set up 4 routes that swap |
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336 channel 0 and channel 1 and leave channel 2 and 3 intact. Observe that if |
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337 media containing two channels was played back, channels 2 and 3 would contain |
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338 silence but 0 and 1 would still be swapped.</P> |
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339 |
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340 <P>Example 2:<BR> |
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341 <CODE>mplayer -af channels=6:4:0:0:0:1:0:2:0:3 media.avi </CODE></P> |
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342 |
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343 <P>would change the number of channels to 6 and set up 4 routes that copy |
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344 channel 0 to channels 0 to 3. Channel 4 and 5 will contain silence.</P> |
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345 |
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346 |
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347 <H5><A NAME="af_format">2.3.2.3.3 Sample format converter</A></H5> |
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348 |
8950 | 349 <P>The <CODE>format</CODE> filter converts between different sample formats. It |
350 is automatically enabled when needed by the sound card or another filter.</P> | |
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351 |
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352 <DL> |
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353 <DT><CODE>bps <number></CODE></DT> |
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354 <DD>can be <CODE>1</CODE>, <CODE>2</CODE> or <CODE>4</CODE> and denotes the |
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355 number of bytes per sample. This switch is required, leaving it empty |
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356 results in a runtime error.</DD> |
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357 |
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358 <DT><CODE>f <format></CODE></DT> |
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359 <DD>is a text string describing the sample format. The string is a |
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360 concatenated mix of: <CODE>alaw</CODE>, <CODE>mulaw</CODE> or |
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361 <CODE>imaadpcm</CODE>, <CODE>float</CODE> or <CODE>int</CODE>, |
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362 <CODE>unsigned</CODE> or <CODE>signed</CODE>, <CODE>le</CODE> or |
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363 <CODE>be</CODE> (little or big endian). This switch is required, leaving it |
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364 empty results in a runtime error.</DD> |
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365 </DL> |
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366 |
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367 <P>Example:<BR> |
8950 | 368 <CODE>mplayer -af format=4:float media.avi</CODE></P> |
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369 |
8772 | 370 <P>would set the output format to 4 bytes per sample floating point |
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371 data.</P> |
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372 |
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373 |
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374 <H5><A NAME="af_delay">2.3.2.3.4 Delay</A></H5> |
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375 |
8950 | 376 <P>The <CODE>delay</CODE> filter delays the sound to the loudspeakers such that |
377 the sound from the different channels arrives at the listening position | |
378 simultaneously. | |
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379 It is only useful if you have more than 2 loudspeakers. This filter has a |
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380 variable number of parameters:</P> |
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381 |
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382 <DL> |
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383 <DT><CODE>d1:d2:d3...</CODE></DT> |
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384 <DD>are floating point numbers representing the delays in ms that should be |
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385 imposed on the different channels. The minimum delay is 0ms and the maximum |
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386 is 1000ms.</DD> |
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387 </DL> |
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388 |
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389 <P>To calculate the required delay for the different channels do as follows:</P> |
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390 |
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391 <OL> |
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392 <LI>Measure the distance to the loudspeakers in meters in relation to your |
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393 listening position, giving you the distances s1 to s5 (for a 5.1 system). |
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394 There is no point in compensating for the sub-woofer (you will not hear the |
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395 difference anyway).</LI> |
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396 <LI>Subtract the distances s1 to s5 from the maximum distance i.e.<BR> |
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397 s[i] = max(s) - s[i]; i = 1...5</LI> |
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398 <LI>Calculated the required delays in ms as<BR> |
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399 d[i] = 1000*s[i]/342; i = 1...5 </LI> |
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400 </OL> |
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401 |
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402 <P>Example:<BR> |
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403 <CODE>mplayer -af delay=10.5:10.5:0:0:7:0 media.avi</CODE></P> |
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404 |
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405 <P>would delay front left and right by 10.5ms, the two rear channels and the sub |
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406 by 0ms and the center channel by 7ms.</P> |
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407 |
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408 |
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409 <H5><A NAME="af_volume">2.3.2.3.5 Software volume control</A></H5> |
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410 |
8950 | 411 <P>Software volume control is implemented by the <CODE>volume</CODE> audio |
412 filter. Use this filter with caution since | |
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413 it can reduce the signal to noise ratio of the sound. In most cases it is best |
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414 to set the level for the PCM sound to max, leave this filter out and control |
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415 the output level to your speakers with the master volume control of the mixer. |
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416 If there is an external amplifier connected to the computer (this is almost |
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417 always the case), the noise level can be minimized by adjusting the master |
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418 level and the volume knob on the amplifier until the hissing noise in the |
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419 background is gone. This filter has two switches:</P> |
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420 |
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421 <DL> |
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422 <DT><CODE>v <-200 - +60></CODE></DT> |
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423 <DD>is a floating point number between <CODE>-200</CODE> and <CODE>+60</CODE> |
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424 which represents the volume level in dB. The default level is -10dB.</DD> |
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425 |
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426 <DT><CODE>c</CODE></DT> |
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427 <DD>is a binary control that turns soft clipping on and off. Soft-clipping can |
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428 make the sound more smooth if very high volume levels are used. Enable this |
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429 switch if the dynamic range of the loudspeakers is very low. Be aware that |
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430 this feature creates distortion and should be considered a last resort.</DD> |
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431 </DL> |
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432 |
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433 <P>Example:<BR> |
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434 <CODE>mplayer -af volume=10.1:0 media.avi</CODE></P> |
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435 |
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436 <P>would amplify the sound by 10.1dB and hard-clip if the sound level is too |
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437 high.</P> |
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438 |
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439 <P>This filter has a second feature: It measures the overall maximum sound level |
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440 and prints out that level when MPlayer exits. This volume estimate can be used |
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441 for setting the sound level in MEncoder such that the maximum dynamic range is |
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442 utilized.</P> |
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443 |
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444 |
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445 <H5><A NAME="af_equalizer">2.3.2.3.6 Equalizer</A></H5> |
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446 |
8950 | 447 <P>The <CODE>equalizer</CODE> filter represents a 10 octave band graphic |
448 equalizer, implemented using 10 IIR | |
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449 band pass filters. This means that it works regardless of what type of audio |
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450 is being played back. The center frequencies for the 10 bands are:</P> |
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451 |
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452 <TABLE BORDER="0" WIDTH="100%"> |
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453 <TR><TD>Band No.</TD><TD>Center frequency</TD></TR> |
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454 <TR><TD>0</TD><TD>31.25 Hz</TD></TR> |
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455 <TR><TD>1</TD><TD>62.50 Hz</TD></TR> |
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456 <TR><TD>2</TD><TD>125.0 Hz</TD></TR> |
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457 <TR><TD>3</TD><TD>250.0 Hz</TD></TR> |
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458 <TR><TD>4</TD><TD>500.0 Hz</TD></TR> |
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459 <TR><TD>5</TD><TD>1.000 kHz</TD></TR> |
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460 <TR><TD>6</TD><TD>2.000 kHz</TD></TR> |
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461 <TR><TD>7</TD><TD>4.000 kHz</TD></TR> |
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462 <TR><TD>8</TD><TD>8.000 kHz</TD></TR> |
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463 <TR><TD>9</TD><TD>16.00 kHz</TD></TR> |
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464 </TABLE> |
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465 |
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466 <P>If the sample rate of the sound being played back is lower than the center |
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467 frequency for a frequency band, then that band will be disabled. A known bug |
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468 with this filter is that the characteristics for the uppermost band are not |
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469 completely symmetric if the sample rate is close to the center frequency of |
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470 that band. This problem can be worked around by up-sampling the sound using |
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471 the resample filter before it reaches this filter. </P> |
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472 |
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473 <P>This filter has 10 parameters:</P> |
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474 |
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475 <DL> |
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476 <DT><CODE>g1:g2:g3...g10</CODE></DT> |
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477 <DD>are floating point numbers between <CODE>-12<CODE> and <CODE>+12</CODE> |
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478 representing the gain in dB for each frequency band.</DD> |
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479 </DL> |
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480 |
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481 <P>Example:<BR> |
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482 <CODE>mplayer -af equalizer=11:11:10:5:0:-12:0:5:12:12 media.avi</CODE></P> |
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483 |
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484 <P>would amplify the sound in the upper and lower frequency region while |
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485 canceling it almost completely around 1kHz.</P> |
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486 |
8950 | 487 |
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488 <H5><A NAME="af_panning">2.3.2.3.7 Panning filter </A></H5> |
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489 |
8950 | 490 <P>Use the <CODE>pan</CODE> filter to mix channels arbitrarily. It is basically |
8750
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491 a combination of the volume control and the channels filter. There are two |
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492 major uses for this filter: </P> |
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493 |
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494 <OL> |
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495 <LI>Down-mixing many channels to only a few, stereo to mono for example.</LI> |
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496 <LI>Varying the "width" of the center speaker in a surround sound system.</LI> |
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497 </OL> |
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498 |
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499 <P>This filter is hard to use, and will require some tinkering before the |
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500 desired result is obtained. The number of switches for this filter depends on |
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501 the number of output channels:</P> |
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502 |
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503 <DL> |
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504 <DT><CODE>nch <1-6></CODE></DT> |
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505 <DD>is an integer between <CODE>1</CODE> and <CODE>6</CODE> and is used for |
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506 setting the number of output channels. This switch is required, leaving it |
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507 empty results in a runtime error.</DD> |
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508 |
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509 <DT><CODE>l00:l01:l02:..l10:l11:l12:...ln0:ln1:ln2:...</CODE></DT> |
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510 <DD>are floating point values between <CODE>0</CODE> and <CODE>1</CODE>. |
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511 <CODE>l[i][j]</CODE> determines how much of input channel j is mixed into |
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512 output channel i.</DD> |
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513 </DL> |
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514 |
8849 | 515 <P>Example 1:<BR> |
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516 <CODE>mplayer -af pan=1:0.5:0.5 -channels 1 media.avi</CODE></P> |
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517 |
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518 <P>would down-mix from stereo to mono.</P> |
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519 |
8849 | 520 <P>Example 2:<BR> |
521 <CODE>mplayer -af pan=3:1:0:1:0.5:0.5 -channels 3 media.avi</CODE></P> | |
522 | |
523 <P>would give 3 channel output leaving channels 0 and 1 intact, and mix channels | |
524 0 and 1 into output channel 2 (which could be sent to a sub-woofer for | |
525 example).</P> | |
526 | |
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527 |
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528 <H5><A NAME="af_sub">2.3.2.3.8 Sub-woofer</A></H5> |
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529 |
8950 | 530 <P>The <CODE>sub</CODE> filter adds a sub woofer channel to the audio stream. |
531 The audio data | |
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532 used for creating the sub-woofer channel is an average of the sound in channel |
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533 0 and channel 1. The resulting sound is then low-pass filtered by a a 4th |
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534 order Butterworth filter with a default cutoff frequency of 60Hz and added to |
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535 a separate channel in the audio stream. Warning: Disable this filter when you |
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536 are playing DVDs with Dolby Digital 5.1 sound, otherwise this filter will |
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537 disrupt the sound to the sub-woofer. This filter has two parameters:</P> |
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538 |
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539 <DL> |
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540 <DT><CODE>fc <20-300></CODE></DT> |
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541 <DD>is an optional floating point number used for setting the cutoff frequency |
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542 for the filter in Hz. The valid range is 20Hz to 300Hz. For the best result |
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543 try setting the cutoff frequency as low as possible. This will improve the |
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544 stereo or surround sound experience. The default cutoff frequency is |
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545 60Hz.</DD> |
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546 |
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547 <DT><CODE>ch <0-5></CODE></DT> |
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548 <DD>is an optional integer between <CODE>0</CODE> and <CODE>5</CODE> which |
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549 determines the channel number in which to insert the sub-channel audio. |
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550 The default is channel number <CODE>5</CODE>. Observe that the number of |
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551 channels will automatically be increased to <CODE>ch</CODE> if |
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552 necessary.</DD> |
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553 </DL> |
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554 |
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555 <P>Example:<BR> |
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556 <CODE>mplayer -af sub=100:4 -channels 5 media.avi</CODE></P> |
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557 |
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558 <P>would add a sub-woofer channel with a cutoff frequency of 100Hz to output |
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559 channel 4.</P> |
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560 |
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561 <H5><A NAME="af_surround">2.3.2.3.9 Surround-sound decoder</A></H5> |
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562 |
8950 | 563 <P>Matrix encoded surround sound can be decoded by the <CODE>surround</CODE> |
564 filter. Dolby Surround is | |
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565 an example of a matrix encoded format. Many files with 2 channel audio |
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566 actually contain matrixed surround sound. To use this feature you need a sound |
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567 card supporting at least 4 channels. This filter has one parameter:</P> |
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568 |
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569 <DL> |
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570 <DT><CODE>d <0-1000></CODE></DT> |
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571 <DD>is an optional floating point number between <CODE>0</CODE> and |
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572 <CODE>1000</CODE> used for setting the delay time in ms for the rear |
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573 speakers. This delay should be set as follows: if d1 is the distance from |
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574 the listening position to the front speakers and d2 is the distance from |
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575 the listening position to the rear speakers, then the delay <CODE>d</CODE> |
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576 should be set to 15ms if d1 <= d2 and to 15 + 5*(d1-d2) if d1 > d2. |
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577 The default value for <CODE>d</CODE> is 20ms.</DD> |
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578 </DL> |
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579 |
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580 <P>Example:<BR> |
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581 <CODE>mplayer -af surround=15 -channels 4 media.avi</CODE></P> |
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582 |
8950 | 583 <P>would add surround sound decoding with 15ms delay for the sound to the rear |
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584 speakers.</P> |
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585 |
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586 |
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587 |
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588 |
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589 <H2><STRONG>Note: Audio plugins have been deprecated by audio filters and will be |
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590 removed soon.</STRONG></H2> |
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591 |
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592 <H4><A NAME="plugins">2.3.2.4 Audio plugins (deprecated)</A></H4> |
3671 | 593 |
7814 | 594 <P>MPlayer has support for audio plugins. Audio plugins can be used for |
4508 | 595 changing the properties of the audio data before the sound reaches the sound |
5765 | 596 card. They are enabled using the <CODE>-aop</CODE> switch which takes a |
597 <CODE>list=plugin1,plugin2,...</CODE> argument. The <CODE>list</CODE> argument | |
598 is required and determines which plugins should be used and in which order they | |
6974 | 599 should be executed. Example:</P> |
4508 | 600 |
601 <P> <CODE>mplayer media.avi -aop list=resample,format</CODE></P> | |
602 | |
603 <P>would run the sound through the resampling plugin followed by the format | |
6974 | 604 plugin.</P> |
4508 | 605 |
5765 | 606 <P>The plugins can also have switches that change their behavior. These |
4508 | 607 switches are explained in detail in the sections below. A plugin will execute |
5765 | 608 using default settings if its switches are omitted. Here is an example of how |
6974 | 609 to use plugins in combination with plugin specific switches:</P> |
4508 | 610 |
611 <P> <CODE>mplayer media.avi -aop | |
6974 | 612 list=resample,format:fout=44100:format=0x8</CODE></P> |
4508 | 613 |
614 <P>would set the output frequency of the resample plugin to 44100Hz and the | |
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615 output format of the format plugin to AFMT_U8.</P> |
4508 | 616 |
7814 | 617 <P>Currently audio plugins can not be used in MEncoder.</P> |
4508 | 618 |
619 | |
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620 <H5><A NAME="resample">2.3.2.4.1 Up/Downsampling</A></H5> |
3671 | 621 |
7814 | 622 <P>MPlayer fully supports up/downsampling of the sound. This plugin can |
5765 | 623 be used if you have a fixed frequency sound card or if you are |
624 stuck with an old sound card that is only capable of max 44.1kHz. | |
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625 Whether is usage of this plugin is necessary or not, is <B>autodetected</B>. |
8260 | 626 This plugin has one switch: |
4508 | 627 <CODE>fout</CODE> which is used for setting the desired output sample |
5765 | 628 frequency. It defaults to 48kHz, and is given in |
6974 | 629 <Hz>.</P> |
3671 | 630 |
5765 | 631 <P>Usage:<BR> |
6974 | 632 <CODE>mplayer media.avi -aop list=resample:fout=<required |
3671 | 633 frequency in Hz, like 44100></CODE></P> |
634 | |
5765 | 635 <P>Note that the output frequency should not be scaled up from the default value. |
636 Scaling up will cause the audio and video streams to be played in slow motion | |
637 in addition to audio distortion.</P> | |
3671 | 638 |
6974 | 639 |
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640 <H5><A NAME="surround_decoding">2.3.2.4.2 Surround Sound decoding</A></H5> |
3931 | 641 |
7814 | 642 <P>MPlayer has an audio plugin that can decode matrix encoded |
6974 | 643 surround sound. Dolby Surround is an example of a matrix encoded format. |
644 Many files with 2 channel audio actually contain matrixed surround sound. | |
645 To use this feature you need a sound card supporting at least 4 channels.</P> | |
3931 | 646 |
5765 | 647 <P>Usage:<BR> |
6974 | 648 <CODE>mplayer media.avi -aop list=surround</CODE></P> |
3931 | 649 |
4508 | 650 |
8177 | 651 <H5><A NAME="format">2.3.2.3.3 Sample format converter</A></H5> |
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652 |
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653 <P>If your sound card driver does not support signed 16bit <CODE>int</CODE> data type, |
5765 | 654 this plugin can |
4508 | 655 be used to change the format to one which your sound card can understand. It |
5765 | 656 has one switch, <CODE>format</CODE>, which can be set to one of the numbers |
657 found in <CODE>libao2/afmt.h</CODE>. This plugin is hardly ever needed and is | |
658 intended for advanced users. Keep in mind that this plugin only changes the | |
6974 | 659 sample format and not the sample frequency or the number of channels.</P> |
4508 | 660 |
5765 | 661 <P>Usage:<BR> |
4508 | 662 <CODE>mplayer media.avi -aop |
6974 | 663 list=format:format=<required output format></CODE></P> |
4508 | 664 |
665 | |
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666 <H5><A NAME="delay">2.3.2.4.4 Delay</A></H5> |
4508 | 667 |
668 <P>This plugin delays the sound and is intended as an example of how to develop | |
5765 | 669 new plugins. It can not be used for anything useful from a users perspective |
4508 | 670 and is mentioned here for the sake of completeness only. Do not use this |
671 plugin unless you are a developer.</P> | |
672 | |
7047 | 673 |
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674 <H5><A NAME="volume">2.3.2.4.5 Software volume control</A></H5> |
4913 | 675 |
676 <P>This plugin is a software replacement for the volume control, and | |
5765 | 677 can be used on machines with a broken mixer device. It can also be |
7814 | 678 used if one wants to change the output volume of MPlayer |
4913 | 679 without changing the PCM volume setting in the mixer. It has one |
680 switch <CODE>volume</CODE> that is used for setting the initial | |
681 sound level. The initial sound level can be set to values between 0 | |
5099 | 682 and 255 and defaults to 101 which equals 0dB amplification. Use this |
683 plugin with caution since it can reduce the signal to noise ratio of | |
684 the sound. In most cases it is best to set the level for the PCM | |
685 sound to max, leave this plugin out and control the output level to | |
5765 | 686 your speakers with the master volume control of the mixer. If there is an |
5099 | 687 external amplifier connected to the computer (this is almost always |
688 the case), the noise level can be minimized by adjusting the master | |
689 level and the volume knob on the amplifier until the hissing noise | |
6974 | 690 in the background is gone.</P> |
4913 | 691 |
5765 | 692 <P>Usage:<BR> |
4913 | 693 <CODE>mplayer media.avi -aop |
6974 | 694 list=volume:volume=<0-255></CODE></P> |
4913 | 695 |
5765 | 696 <P>This plugin also has compressor or "soft-clipping" capabilities. |
5099 | 697 Compression can be used if the dynamic range of the sound is very |
698 high or if the dynamic range of the loudspeakers is very | |
5765 | 699 low. Be aware that this feature creates distortion and should be |
6974 | 700 considered a last resort.</P> |
5099 | 701 |
5765 | 702 <P>Usage:<BR> |
5099 | 703 <CODE>mplayer media.avi -aop |
6974 | 704 list=volume:softclip</CODE></P> |
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705 |
5109 | 706 |
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707 <H5><A NAME="extrastereo">2.3.2.4.6 Extrastereo</A></H5> |
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708 |
5765 | 709 <P>This plugin (linearly) increases the difference between left and right |
710 channels (like the XMMS extrastereo plugin) which gives some sort of "live" | |
6974 | 711 effect to playback.</P> |
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712 |
5765 | 713 <P>Usage:<BR> |
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714 <CODE>mplayer media.avi -aop list=extrastereo</CODE><BR> |
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715 <CODE>mplayer media.avi -aop list=extrastereo:mul=3.45</CODE></P> |
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716 |
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717 <P>The default coefficient (<CODE>mul</CODE>) is a float number that defaults |
5765 | 718 to 2.5. If you set it to 0.0, you will have mono sound (average of both |
6272 | 719 channels). If you set it to 1.0, sound will be unchanged, if you set it to |
720 -1.0, left and right channels will be swapped.</P> | |
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721 |
5109 | 722 |
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723 <H5><A NAME="normalizer">2.3.2.4.7 Volume normalizer</A></H5> |
5109 | 724 |
5765 | 725 <P>This plugin maximizes the volume without distorting the sound.</P> |
5109 | 726 |
5765 | 727 <P>Usage:<BR> |
5109 | 728 <CODE>mplayer media.avi -aop list=volnorm</CODE><BR> |
729 | |
5721 | 730 |
1612 | 731 </BODY> |
732 </HTML> |