Mercurial > mplayer.hg
annotate DOCS/German/sound.html @ 8846:e293d5eccd41
bunkus: updated translations. removed the boldness from <B>MPlayer</B>. removed DOS newlines (yuck).
author | mosu |
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date | Wed, 08 Jan 2003 22:04:40 +0000 |
parents | e4c5ee3aa3e9 |
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1 <!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> |
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2 <HTML> |
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4 <HEAD> | |
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5 <TITLE>Sound - MPlayer - The Movie Player for Linux</TITLE> |
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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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9 | |
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10 <BODY> |
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11 |
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12 <H3><A NAME="audio">2.3.2. Audio-Ausgabe-Geräte</A></H3> |
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13 |
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14 <H4><A NAME="sync">2.3.2.1 Audio-/VideoSynchronisation</A></H4> |
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15 |
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16 <P>MPlayers Audio-Interface wird <I>libao2</I> genannt. Es enthält |
1959 | 17 zur Zeit diese Treiber:</P> |
18 | |
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19 <DL> |
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20 <DT>oss</DT> |
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21 <DD>OSS (ioctl)-Treiber (unterstützt das Durchreichen von AC3)</DD> |
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22 |
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23 <DT>sdl</DT> |
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24 <DD>SDL-Treiber (untertstützt <B>ESD</B>, <B>ARTS</B> etc)</DD> |
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25 |
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26 <DT>nas</DT> |
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27 <DD>NAS (Network Audio System)-Treiber</DD> |
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28 |
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29 <DT>alsa5</DT> |
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30 <DD>ALSA 0.5-Treiber</DD> |
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31 |
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32 <DT>alsa9</DT> |
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33 <DD>ALSA 0.9-Treiber (unterstützt das Durchreichen von AC3)</DD> |
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34 |
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35 <DT>sun</DT> |
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36 <DD>SUN Audiotreiber (<CODE>/dev/audio</CODE>) für BSD und Solaris8</DD> |
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37 |
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38 <DT>arts</DT> |
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39 <DD>ARTS-Treiber (hauptsächlich für KDE-Benutzer)</DD> |
1959 | 40 |
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41 <DT>esd</DT> |
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42 <DD>ESD-Treiber (hauptsächlich für GNOME-Benutzer)</DD> |
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43 </DL> |
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44 |
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45 <P>Fakt ist, dass Linux-Soundkartentreiber Kompatibilitätsprobleme haben. |
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46 Der Grund für diese Probleme liegt darin, dass MPlayer ein Feature |
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47 des Soundkartentreibers benutzt, um Audio und Video synchron zu halten. Leider |
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48 kümmern sich einige Soundkartentreiberprogrammierer nicht besonders um |
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49 diese Funktion, da sie weder für die Wiedergabe von MP3s noch für |
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50 Soundeffeket benötigt wird.</P> |
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51 |
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52 <P>Andere Medienplayer wie z.B. aviplay oder xine funktionieren bei dir |
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53 wahrscheinlich so, wie sie sind, weil sie eine "simplere" Methode wie internes |
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54 Timing für die Audio-/Video-Synchronisation verwenden. Anmerkung: es hat |
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55 sich oft gezeigt, dass ihre Methoden nicht ganz so effizient wie |
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56 MPlayers Methoden sind.</P> |
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57 |
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58 <P>Wenn du MPlayer zusammen mit einem korrekt geschriebenen |
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59 Soundkartentreiber verwendest, dann wirst du niemals A/V-Desyncs aufgrund des |
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60 Audiotreibers erleben (höchstens mit sehr schlecht erstellten Dateien. |
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61 Schau in einem solchen Fall in die Dokumentation, wie du das beheben |
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62 kannst).</P> |
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63 |
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64 <P>Einige Anmerkungen:</P> |
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65 |
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66 <UL> |
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67 <LI>Wenn du einen OSS-Treiber hast, versuche als erstes <CODE>-ao |
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68 oss</CODE> (dies ist die Standardeinstellung). Wenn du kleine Fehler, |
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69 Verzögerungen oder irgendetwas anderes ungewöhliches hörst, |
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70 versuche mal <CODE>-ao sdl</CODE> (HINWEIS: die SDL-Bibliotheken und das |
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71 dazugehörige Devel-Paket müssen installiert sein). Der SDL- |
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72 Audiotreiber hilft in vielen Fällen (er unterstützt auch ESD, |
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73 ARTS und Up-/Downsampling. ESD ist der Sounddaemon von GNOME, arts der |
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74 von KDE).</LI> |
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75 <LI>Wenn du ALSA Version 0.5 verwendest, musst du fast immer <CODE>-ao |
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76 alsa5</CODE> verwenden, weil ALSA 0.5 einen fehlerhaften OSS- |
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77 Emulationscode hat. MPlayer wird mit etwa so einer Meldung |
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78 <B>abstürzen</B>:<BR> <CODE>DEMUXER: Too many (945 in 8390980 bytes) |
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79 video packets in the buffer!</CODE></LI> |
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80 </UL> |
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81 |
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82 <P>Verwende auf <B>Solaris</B>-Systemen den SUN-Audio-Treiber mit |
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83 <CODE>-ao sun</CODE>, ansonsten wirst du weder Video noch Ton haben.</P> |
1959 | 84 |
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85 |
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86 <H4><A NAME="experiences">2.3.2.2. Soundkarten-Erfahrungen, |
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87 -Empfehlungen</A></H4> |
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88 |
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89 <TABLE BORDER="0" WIDTH="100%"> |
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90 <TR><TD COLSPAN=3><B>VIA Onboard-Chipsatz (via82cxxx), nur 48kHz</B></TD></TR> |
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91 <TR><TD></TD><TD>Treiber:</TD> |
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92 <TD>von <A HREF="http://sourceforge.net/project/showfiles.php?group_id=3242&release_id=59602">sourceforge.net</A></TD></TR> |
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93 |
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94 <TR><TD COLSPAN=3><B>Aureal Vortex 2</B></TD></TR> |
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95 <TR><TD> </TD><TD>OSS:</TD><TD>kein Treiber</TD></TR> |
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96 <TR><TD></TD><TD>OSS/Pro:</TD><TD>OK</TD></TR> |
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97 <TR><TD></TD><TD>ALSA:</TD><TD>kein Treiberr</TD></TR> |
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98 <TR><TD></TD><TD>Max kHz:</TD><TD>48</TD></TR> |
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99 <TR><TD></TD><TD>Treiber:</TD> |
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100 <TD><A HREF="http://aureal.sourceforge.net">aureal.sourceforge.net</A></TD></TR> |
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101 <TR><TD></TD><TD>Treiber 2:</TD> |
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102 <TD>von <A HREF="http://makacs.poliod.hu/~pontscho/aureal/au88xx-1.1.3.tar.bz2">Pontscho's Seite</A><BR> |
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103 (<I>Puffergröße auf 32k erhöht</I>)</TD></TR> |
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104 |
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105 <TR><TD COLSPAN=3><B>GUS PnP</B></TD></TR> |
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106 <TR><TD></TD><TD>OSS:</TD><TD>kein Treiber</TD></TR> |
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107 <TR><TD></TD><TD>OSS/Pro:</TD><TD>OK</TD></TR> |
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108 <TR><TD></TD><TD>ALSA:</TD><TD>OK</TD></TR> |
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109 <TR><TD></TD><TD>Max kHz:</TD><TD>48</TD></TR> |
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110 |
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111 <TR><TD COLSPAN=3><B>SB Live!</B></TD></TR> |
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112 <TR><TD></TD><TD>OSS:</TD><TD>Analog OK, SP/DIF funktioniert nicht</TD></TR> |
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113 <TR><TD></TD><TD>ALSA:</TD><TD>beide OK</TD></TR> |
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114 <TR><TD></TD><TD>Max kHz:</TD><TD>192</TD></TR> |
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115 |
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116 <TR><TD COLSPAN=3><B>SB AWE 64</B></TD></TR> |
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117 <TR><TD></TD><TD>OSS:</TD><TD>max 44kHz</TD></TR> |
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118 <TR><TD></TD><TD>ALSA:</TD><TD>48kHz klingt schlecht</TD></TR> |
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119 <TR><TD></TD><TD>Max kHz:</TD><TD>48</TD></TR> |
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120 |
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121 <TR><TD COLSPAN=3><B>Gravis UltraSound ACE</B></TD></TR> |
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122 <TR><TD></TD><TD>OSS:</TD><TD>nicht OK</TD></TR> |
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123 <TR><TD></TD><TD>ALSA:</TD><TD>OK</TD></TR> |
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124 <TR><TD></TD><TD>Max kHz:</TD><TD>44</TD></TR> |
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125 |
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126 <TR><TD COLSPAN=3><B>Gravis UltraSound MAX</B></TD></TR> |
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127 <TR><TD></TD><TD>OSS:</TD><TD>OK</TD></TR> |
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128 <TR><TD></TD><TD>ALSA:</TD><TD>OK (?)</TD></TR> |
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129 <TR><TD></TD><TD>Max kHz:</TD><TD>48</TD></TR> |
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130 |
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131 <TR><TD COLSPAN=3><B>ESS 688</B></TD></TR> |
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132 <TR><TD></TD><TD>OSS:</TD><TD>OK</TD></TR> |
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133 <TR><TD></TD><TD>ALSA:</TD><TD>OK (?)</TD></TR> |
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134 <TR><TD></TD><TD>Max kHz:</TD><TD>48</TD></TR> |
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135 |
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136 <TR><TD COLSPAN=3><B>C-Media-Karten (welche genau?)</B></TD></TR> |
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137 <TR><TD></TD><TD>OSS:</TD><TD>nicht OK (scharfe ssss-Geräusche) (?)</TD></TR> |
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138 <TR><TD></TD><TD>ALSA:</TD><TD>OK (?)</TD></TR> |
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139 <TR><TD></TD><TD>Max kHz:</TD><TD>?</TD></TR> |
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140 |
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141 <TR><TD COLSPAN=3><B>Yamaha-Karten (*ymf*)</B></TD></TR> |
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142 <TR><TD></TD><TD>OSS:</TD><TD>nicht OK (?) (evtl. <CODE>-ao sdl</CODE>)</TD></TR> |
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143 <TR><TD></TD><TD>ALSA:</TD><TD>nur mit der OSS-Emulation von ALSA 0.5 |
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144 <B>UND</B> <CODE>-ao sdl</CODE> OK (!) (?)</TD></TR> |
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145 <TR><TD></TD><TD>Max kHz:</TD><TD>?</TD></TR> |
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146 |
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147 <TR><TD COLSPAN=3><B>Karten mit envy24-Chips (wie z.B. Terratec EWS88MT)</B></TD></TR> |
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148 <TR><TD></TD><TD>OSS:</TD><TD>?</TD></TR> |
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149 <TR><TD></TD><TD>OSS/Pro:</TD><TD>OK</TD></TR> |
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150 <TR><TD></TD><TD>ALSA:</TD><TD>?</TD></TR> |
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151 <TR><TD></TD><TD>Max kHz:</TD><TD>?</TD></TR> |
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152 |
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153 <TR><TD COLSPAN=3><B>PC-Lautsprecher oder DAC</B></TD></TR> |
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154 <TR><TD></TD><TD>OSS:</TD><TD>OK (Benutz den SDL-Treiber: <CODE>-ao sdl</CODE>)</TD></TR> |
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155 <TR><TD></TD><TD>ALSA:</TD><TD>kein Treiber</TD></TR> |
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156 <TR><TD></TD><TD>Max kHz:</TD><TD>Der Treiber emuliert 44.1, evtl auch mehr.</TD></TR> |
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157 <TR><TD></TD><TD>Treiber:</TD><TD><A HREF="ftp://ftp.infradead.org/pub/pcsp">ftp://ftp.infradead.org/pub/pcsp</A></TD></TR> |
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158 </TABLE> |
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159 |
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160 <P>Unter Linux wird ein 2.4.x Kernel sehr empfohlen. Kernel 2.2 wird nicht |
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161 getestet.</P> |
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162 |
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163 <P>Wenn der sound während der Wiedergabe von CD-ROM "klickende" |
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164 Geräusche verursacht, schalte die Interupt-Unmaskierung ein, zum Beispiel |
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165 <CODE>hdparm -u1 /dev/cdrom</CODE> (<CODE>man hdparm</CODE>). Dies ist |
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166 allgemein besser und wird in der <A HREF="cd-dvd.html#drives">CD-ROM- |
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167 Sektion</A> genauer beschrieben.</P> |
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168 |
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169 <P>Rückmeldungen zu diesem Dokument sind willkommen. Bitte sag uns, wie |
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170 MPlayer und deine Soundkarte(n) zusammen funktionieren.</P> |
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171 |
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172 <H4><A NAME="af">2.3.2.3 Audiofilter</A></H4> |
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173 |
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174 <P>Die alten Audioplugins wurden von einer neuen Filterschicht abgelöst. |
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175 Audiofilter werden benutzt, um die Parameter der Daten zu ändern, bevor |
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176 sie die Soundkarte erreichen. Die Aktivierung der Filter geschieht normalerweise |
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177 automatisch, kann aber auch erzwungen oder verhindert werden. Die Filter |
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178 werden aktiviert, wenn die Parameter der Daten von denen der Soundkarte |
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179 abweichen und deaktiviert, wenn sie gleich sind. Die Option <CODE>-af |
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180 filter1,filter2,...</CODE> wird benutzt, um Filter einzufügen, die nicht |
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181 automatisch eingefügt werden. Diese Filter werden in der Reihenfolge |
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182 abgearbeitet, wie sie angegeben wurden.</P> |
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183 |
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184 <P>Beispiele:<BR> |
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185 <CODE>mplayer -af resample,pan movie.avi </CODE></P> |
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186 |
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187 <P>Schickt die Audiodaten durch das Resample-Filter und danach durch den |
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188 Pan-Filter. Die Liste darf keine Leerzeichen enthalten, sonst klappt's nicht. |
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189 </P> |
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190 |
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191 <P>Viele Filter kennen Parameter, mit denen ihr Verhalten geändert werden |
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192 kann. Diese Parameter werden unten detailliert beschrieben. Wenn keine |
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193 Parameter angegeben werden, dann verwendet der Filter seine Standardwerte. Hier |
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194 ist ein Beispiel, wie Filter mit Parametern verwendet werden können:</P> |
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195 |
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196 <P> <CODE>mplayer -af resample=11025,pan=1:0.5:0.5 -channels 1 |
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197 -srate 11025 media.avi</CODE></P> |
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198 |
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199 <P>Das setzt die Ausgabefrequenz des Resampleplugins auf 11025Hz und |
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200 downmixt das Audio auf einen Kanal mit dem Pan-Filter.</P> |
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201 |
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202 <P>Die meisten Filter geben mehr Statusinformationen aus, wenn <CODE>-v</CODE> |
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203 benutzt wird.</P> |
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204 |
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205 <P>Die allgemeine Kontrolle der Filterschicht geschieht über |
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206 <CODE>-af-adv</CODE>. Diese Option kennt zwei Parameter:</P> |
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207 |
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208 <DL> |
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209 <DT><CODE>force</CODE><DT> |
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210 <DD>ist eine ganze Zahl zwischen 0 und 3, die angibt, wie die Filter |
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211 einzufügen sind und welche Geschwindigkeits-/Genauigkeitsoptimierungen |
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212 sie verwenden: |
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213 <DL> |
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214 <DT>0</DT> |
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215 <DD>Benutzt automatisches Einfügen und optimiert in Abhängigkeit |
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216 der CPU-Geschwindigkeit.</DD> |
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217 <DT>1</DT> |
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218 <DD>Benutzt automatisches Einfügen und optimiert für die |
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219 höchstmögliche Geschwindigkeit. Wenn diese Option gesetzt |
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220 ist, dann wird Fixkommaarithmetik verwendet. Warnung: Einige |
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221 Features der Audiofilter werden kommentarlos deaktiviert, und die |
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222 Soundqualität könnte niedriger sein als erwartet.</DD> |
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223 <DT>2</DT> |
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224 <DD>Benutzt automatisches Einfügen der Filter und optimiert für |
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225 Qualität. Wenn diese Option gesetzt ist, dann wird |
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226 Fließkommaarithmetik verwendet. Das ist natürlich sehr |
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227 CPU-intensiv, liefert aber auch die beste Qualität.</DD> |
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228 <DT>3</DT> |
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229 <DD>Füt nicht automatisch Filter ein. Warnung: Hiermit kann |
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230 MPlayer eventuell zum Absturz gebracht werden.</DD> |
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231 </DL> |
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232 </DD> |
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233 |
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234 <DT><CODE>list</CODE></DT> |
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235 <DD>Dies ist ein Synonym für <CODE>-af</CODE>.</DD> |
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236 </DL> |
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237 |
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238 |
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239 <H5><A NAME="af_resample">2.3.2.3.1 Up/Down-sampling</A></H5> |
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240 |
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241 <P>MPlayer unterstützt vollständig Up-/Downsampling. Dieser Filter |
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242 kann benutzt werden, wenn du eine Soundkarte hast, die nur eine feste Frequenz |
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243 unterstützt, oder die nur maximal 44.1kHz unterstützt. Dieser Filter |
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244 wird automatisch aktiviert, wenn er benötigt wird. Er kann aber auch |
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245 explizit auf der Kommandozeile aktiviert werden. Er kennt drei Optionen:</P> |
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246 |
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247 <DL> |
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248 <DT><CODE>srate</CODE></DT> |
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249 <DD>wird benutzt, um die Ausgabesmaplefrequenz in Hz anzugeben. Der |
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250 gültige Bereich liegt zwischen 8kHz und 192kHz. Wenn Eingangs- |
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251 und Ausgabefrequenz gleich sind, oder wenn dieser Parameter nicht angegeben |
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252 wurde, so wird der Filter wieder entladen. Hohe Samplefrequenzen verbessern |
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253 normalerweise die Audioqualität, vor allem, wenn auch andere |
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254 Filter benutzt werden.</DD> |
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255 |
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256 <DT><CODE>sloppy</CODE></DT> |
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257 <DD>Mit dieser Option wird zugelassen, dass die Ausgabefrequenz leicht von |
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258 der geforderten Frequenz (<CODE>srate</CODE>) abweicht. Diese Option |
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259 kann benutzt werden, wenn das Playback nur sehr langsam startet.</DD> |
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260 |
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261 <DT><CODE>fast</CODE><DT> |
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262 <DD>Dieser Parameter aktiviert otpional lineare Interpolation als |
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263 Resamplingmethode. Lineare Interpolation ist sehr schnell, liefert aber |
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264 auch schlechtere Soundqualität, vor allem in Verbindung mit |
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265 Up-/Downsampling.</DD> |
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266 </DL> |
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267 |
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268 <P>Beispiele:<BR> |
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269 <CODE>mplayer -af resample=44100:0:1</CODE></P> |
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270 |
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271 <P>Dies setzt die Ausgabefrequenz des Resample-Filters auf exakt 44100Hz |
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272 mit linearer Interpolation.</P> |
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273 |
3707 | 274 |
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275 <H5><A NAME="af_channels">2.3.2.3.2 Ändern der Anzahl der Kanäle</A> |
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276 </H5> |
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277 |
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278 <P>Der <CODE>channels</CODE>-Filter kann benutzt werden, um Kanäle |
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279 hinzuzufügen oder zu entfernen. Er kann auch benutzt werden, um |
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280 Kanäle umzuleiten oder sie zu kopieren. Er wird automatisch aktiviert, |
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281 falls die Anzahl der Kanäle bei Eingang und Ausgang der Audiofilterschicht |
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282 unterschiedlich ist, oder wenn ein anderer Filter diesen Filter benötigt. |
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283 Wenn er nicht gebraucht wird, so entlädt sich dieser Filter automatisch. |
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284 Die Anzahl der Parameter ist dynamisch:</P> |
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285 |
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286 <DL> |
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287 <DT><CODE>nch</CODE></DT> |
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288 <DD>ist eine ganze Zahl zwischen 1 und 6, die die Nummer der |
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289 Ausgabekanäle festlegt. Dieser Parameter wird benötigt. Wenn er |
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290 weggelassen wird, dann erhält man einen Laufzeitfehler.</DD> |
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291 |
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292 <DT><CODE>nr</CODE></DT> |
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293 <DD>ist eine ganze Zahl zwischen 1 und 6, die die Anzahl der Umleitungen |
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294 festlegt. Dieser Parameter ist optional. Wenn er weggelassen wird, dann |
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295 werden keine Umleitungen vorgenommen.</DD> |
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296 |
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297 <DT><CODE>von1:nach1:von2:nach2:von3:nach3...</CODE></DT> |
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298 <DD>sind paare von Nummern zwischen 0 und 5, die festlegen, wohin jeder |
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299 Kanal umgeleitet werden soll.</DD> |
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300 </DL> |
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301 |
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302 <P>Wenn nur <CODE>nch</CODE> angegeben wird, dann werden die |
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303 Standardumleitungen benutzt, die so aussehen: Wenn mehr Ausgabekanäle |
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304 gefordert sind als Eingangskanäle existieren, so werden leere Kanäle |
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305 erzeugt (bis auf's Mixen von Mono auf Stereo: hier wird der Monokanal auf beide |
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306 Kanäle kopiert). Ist die Anzahl dagegen kleiner, so werden die |
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307 überschüssigen Kanäle entfernt.</P> |
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308 |
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309 <P>Beispiel 1:<BR> |
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310 <CODE>mplayer -af channels=4:4:0:1:1:0:2:2:3:3 media.avi </CODE></P> |
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311 |
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312 <P>Ändert die Anzahl der Kanäle auf 4 und legt vier Routen fest, die |
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313 die Kanäle 0 und 1 vertauschen und die Kanäle 2 und 3 intakt lassen. |
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314 Wenn die Quelldatei nur zwei Kanäle hat, so sind die Kanäle 2 und 3 |
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315 leise, aber 0 und 1 werden trotzdem vertauscht.</P> |
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316 |
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317 <P>Beispiel 2:<BR> |
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318 <CODE>mplayer -af channels=6:4:0:0:0:1:0:2:0:3 media.avi </CODE></P> |
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319 |
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320 <P>Ändert die Anzahl der Kanäle auf 6 und gibt 4 Routen an, die Kanal |
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321 0 auf Kanal 0 und 3 kopieren. Die Kanäle 4 und 5 bleiben stumm. </P> |
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322 |
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323 <H5><A NAME="af_format">2.3.2.3.3 Formatkonvertierung</A></H5> |
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324 |
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325 <P>Dieser Filter konviert das Sampleformat. Er wird automatisch aktiviert, |
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326 wenn er benötigt wird.</P> |
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327 |
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328 <DL> |
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329 <DT><CODE>bps</CODE></DT> |
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330 <DD>kann 1, 2 oder 4 sein und gibt die Anzahl der Bytes pro Sample an. |
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331 Dieser Parameter wird benötigt und führt zu einem Laufzeitfehler, |
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332 wenn er weggelassen wird.</DD> |
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333 |
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334 <DT><CODE>f</CODE></DT> |
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335 <DD>ist ein Textstring, der das Sampleformat beschreibt. Der String besteht |
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336 aus einem Mix der folgenden Schlüsselwörter: |
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337 <CODE>alaw</CODE>, <CODE>mulaw</CODE> oder |
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338 <CODE>imaadpcm</CODE>, <CODE>float</CODE> oder <CODE>int</CODE>, |
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339 <CODE>unsigned</CODE> oder <CODE>signed</CODE>, <CODE>le</CODE> oder |
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340 <CODE>be</CODE> (little oder big endian). |
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341 Dieser Parameter wird benötigt und führt zu einem Laufzeitfehler, |
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342 wenn er weggelassen wird.</DD> |
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343 </DL> |
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344 |
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345 <P>Beispiel:<BR> |
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346 <CODE>mplayer media.avi -af format=4:float</CODE></P> |
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347 |
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348 <P>setzt das Ausgabeformat auf 4 Bytes pro Sample Fließkommadaten.</P> |
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349 |
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350 |
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351 <H5><A NAME="af_delay">2.3.2.3.4 Verzögerung</A></H5> |
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352 |
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353 <P>Dieser Filter verzögert den Sound auf dem Weg zum Lautsprecher, damit |
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354 die verschiedenen Kanäle zur gleichen Zeit bei der Sitzposition ankommen. |
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355 Dieser Filter ist nur dann nützlich, wenn du mehr als zwei Lautsprecher |
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356 hast. Dieser Filter erhält eine variable Anzahl Parameter:</P> |
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357 |
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358 <DL> |
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359 <DT><CODE>d1:d2:d3...</CODE></DT> |
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360 <DD>sind Fließkommazahlen, die die Verzögerung in ms für |
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361 den jeweiligen Kanal angeben. Minimum ist 0ms, das Maximum 1000ms.</DD> |
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362 </DL> |
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363 |
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364 <P>Um die benötigte Verzögerung für die verschiedenen |
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365 Kanäle zu berechnen, tu folgendes:</P> |
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366 |
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367 <OL> |
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368 <LI>Miss die Entfernung zwischen den Lautsprechern und deiner Sitzposition |
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369 in Metern. Dies gibt dir die Distanzen s1 bis s5 (bei einem 5.1-System). |
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370 Es lohnt sich nicht, auch für den Subwoofer zu kompensieren, weil |
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371 man den Unterschied eh nicht hört.</LI> |
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372 <LI>Subtrahiere die Distanzen s1 bis s5 von der maximalen Distanz, z.B.<BR> |
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373 s[i] = max(s) - s[i]; i = 1...5</LI> |
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374 <LI>Berechne die benötigte Verzögerung wie folgt:<BR> |
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375 d[i] = 1000*s[i]/342; i = 1...5 </LI> |
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376 </OL> |
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377 |
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378 <P>Beispiel:<BR> |
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379 <CODE>mplayer -af delay=10.5:10.5:0:0:7:0 media.avi</CODE></P> |
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380 |
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381 <P>verzögert die Kanäle vorne links und vorne rechts um 10.5ms, |
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382 die beiden hinteren Kanäle um 0ms und den zentrierten Kanal um 7ms.</P> |
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383 |
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384 |
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385 <H5><A NAME="af_volume">2.3.2.3.5 Lautstärkekontrolle mit Software</A></H5> |
3707 | 386 |
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387 <P>Dieser Filter stellt eine Softwarelautstärkeregelung dar. Sei bei der |
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388 Benutzung dieses Filters vorsichtig, da er den Signal-zu-Rausch-Abstand |
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389 verringern kann. In den meisten Fällen ist es besser, beim Mixer deiner |
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390 Soundkarte den PCM-Regler auf's Maximum zu setzen und diesen Filter |
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391 wegzulassen. Ändere dann die Lautstärke mit dem Hauptregler des |
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392 Mixers. Wenn der Computer an einen externen Verstärker angeschlossen ist |
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393 (was meistens der Fall ist), dann kann der Rauschpegel minimiert werden, indem |
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394 der Hauptregler und der Lautstärkeregler des Verstärkers angepasst |
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395 werden, bis das Hintergrundzischen verschwunden ist. Der Filter selber kennt |
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396 zwei Parameter:</P> |
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397 |
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398 <DL> |
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399 <DT><CODE>v</CODE></DT> |
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400 <DD>ist eine Fließkommazahl zwischen -200 und +60 und gibt die |
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401 Lautstärke in dB an. Der Standardlevel ist -10dB.</DD> |
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402 |
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403 <DT><CODE>c</CODE></DT> |
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404 <DD>ist ein Binärwert, der soft clipping an- oder ausschaltet. |
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405 Soft clipping kann den Sound glätten, wenn sehr laute Passagen |
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406 auftreten. Aktivier diese Option, wenn die Dynamikbandbreite deiner |
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407 Lautsprecher gering ist. Sei gewanrt, dass dieses Feature Verzerrungen |
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408 hervorruft. Es sollte nur als letzte Möglichkeit benutzt werden.</DD> |
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409 </DL> |
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410 |
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411 <P>Beispiel:<BR> |
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412 <CODE>mplayer -af volume=10.1:0 media.avi</CODE></P> |
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413 |
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414 <P>verstärkt den Sound um 10.1dB und schneidet hart ab, wenn die |
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415 Lautstärke zu hoch wird.</P> |
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416 |
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417 <P>Dieser Filter besitzt noch ein zweites Feature: Er misst die maximale |
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418 Lautstärke und gibt sie aus, wenn MPlayer beendet wird. Dieser Wert |
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419 kann benutzt werden, um die Verstärkung in MEncoder so zu setzen, dass |
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420 die Dynamikbandbreite vollständig ausgenutzt wird.</P> |
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421 |
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422 |
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423 <H5><A NAME="af_equalizer">2.3.2.3.6 Equalizer</A></H5> |
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424 |
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425 <P>Dieser Filter ist ein 10-Band Graphicequalizer, der durch 10 |
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426 IIR-Bandpassfilter (infinite impulse response, rekursive Filter) realisiert |
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427 wird. Das bedeutet, dass er unabhängig vom wiederzugebenden Audiotyp |
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428 funktioniert. Die Mittenfrequenzen der zehn Bändern sind:</P> |
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429 |
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430 <TABLE BORDER="0" WIDTH="100%"> |
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431 <TR><TD>Band Nr.</TD><TD>Mittenfrequenz</TD></TR> |
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432 <TR><TD>0</TD><TD>31.25 Hz</TD></TR> |
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433 <TR><TD>1</TD><TD>62.50 Hz</TD></TR> |
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434 <TR><TD>2</TD><TD>125.0 Hz</TD></TR> |
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435 <TR><TD>3</TD><TD>250.0 Hz</TD></TR> |
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436 <TR><TD>4</TD><TD>500.0 Hz</TD></TR> |
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437 <TR><TD>5</TD><TD>1.000 kHz</TD></TR> |
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438 <TR><TD>6</TD><TD>2.000 kHz</TD></TR> |
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439 <TR><TD>7</TD><TD>4.000 kHz</TD></TR> |
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440 <TR><TD>8</TD><TD>8.000 kHz</TD></TR> |
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441 <TR><TD>9</TD><TD>16.00 kHz</TD></TR> |
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442 </TABLE> |
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443 |
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444 <P>Wenn die Samplefrequenz des Sounds niedriger als die Mittenfrequenz eines |
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445 Bandes ist, so wird dieses Band deaktiviert. Ein bekannter Fehler dieses |
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446 Filters liegt darin, dass die Charakteristiken des höchsten Bandes nicht |
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447 ganz symmetrisch sind, wenn die Samplefrequenz nahe der Mittenfrequenz des |
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448 Bandes liegt. Das kann umgangen werden, wenn vorher ein Upsampling mit dem |
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449 resample-Filter vorgenommen wird.</P> |
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450 |
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451 <P>Dieser Filter bekommt zehn Parameter:</P> |
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452 |
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453 <DL> |
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454 <DT><CODE>g1:g2:g3...g10</CODE></DT> |
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455 <DD>sind Fließkommazahlen zwischen -12 und +12dB, die die |
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456 Verstärkung in dB des jeweiligen Bandes angeben.</DD> |
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457 </DL> |
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458 |
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459 <P>Beispiel:<BR> |
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460 <CODE>mplayer -af equalizer=11:11:10:5:0:-12:0:5:12:12 media.avi</CODE></P> |
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461 |
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462 <P>verstärkt den Sound in den oberen und unteren Frequenzbereichen und |
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463 löscht ihn um 1kHz beinahe komplett aus.</P> |
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464 |
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465 <H5><A NAME="af_panning">2.3.2.3.7 Panning-Filter </A></H5> |
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466 |
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467 <P>Dieser Filter kann benutzt werden, um Kanäle beliebig zu mischen. Er |
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468 ist grundlegend nur eine Kombination des Lautstärkefilters und des |
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469 channels-Filters. Es gibt zwei Haupteinsatzgebiete für diesen Filter:</P> |
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470 |
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471 <OL> |
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472 <LI>Heruntermischen vieler Kanäle auf wenige, z.B. Stereo zu Mono.</LI> |
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473 <LI>Verändern der "Breite" des zentrierten Kanals bei |
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474 Sourround-Sound-Systemem.</LI> |
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475 </OL> |
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476 |
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477 <P>Dieser Filter ist nicht ganz leicht zu bedienen und benötigt viel |
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478 Ausprobieren, bevor die gewünschten Ergebnisse erzielt werden. Die Anzahl |
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479 der Parameter hängt von der Anzahl der Ausgabekanäle ab:</P> |
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480 |
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481 <DL> |
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482 <DT><CODE>nch</CODE></DT> |
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483 <DD>ist eine ganze Zahl zwischen 1 und 6 und wird benutzt, um die Anzahl |
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484 der Ausgabekanäle zu setzen. Dieser Parameter wird benötigt. |
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485 Wenn er weggelassen wird, erhält man einen Laufzeitfehler.</DD> |
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486 <DT><CODE>l00:l01:l02:..l10:l11:l12:...ln0:ln1:ln2:...</CODE></DT> |
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487 <DD>sind Fließkommazhalen zwischen 0 und 1. <CODE>l[i][j]</CODE> |
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488 gibt an, wieviel vom Einganskanal j in den Ausgabekanal i zu mischen |
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489 ist.</DD> |
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490 </DL> |
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491 |
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492 <P>Beispiel:<BR> |
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493 <CODE>mplayer -af pan=1:0.5:0.5 -channels 1 media.avi</CODE></P> |
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494 |
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495 <P>mischt stereo auf mono.</P> |
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496 |
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497 |
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498 <H2><STRONG>Anmerkung: Audioplugins sind veraltet und wurden durch Audiofilter |
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499 ersetzt. Sie werden bald entfernt werden.</STRONG></H2> |
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500 |
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501 <H4><A NAME="plugins">2.3.2.4. Audio-Plugins (veraltet)</A></H4> |
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502 |
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503 <P>MPlayer bietet Unterstützung für Audio-Plugins. Audio- |
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504 Plugins können benutzt werden, um die Einstellungen der Audio-Daten zu |
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505 ändern, bevor sie die Soundkarte erreichen. Sie aktiviert durch die |
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506 Option <CODE>-aop</CODE> aktiviert, welche |
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507 <CODE>list=plugin1,plugin2,...</CODE> als Argument hat. Das <CODE>list</CODE> |
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508 wird benötigt, um festzulegen, in welcher Reihenfolge die Plugins benutzt |
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509 werden sollen. Beispiel:</P> |
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510 |
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511 <P> <CODE>mplayer media.avi -aop list=resample,format</CODE></P> |
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512 |
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513 <P>würde den Ton durch das Resampling-Plugin laufen lassen, gefolgt vom |
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514 Format-Plugins.</P> |
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515 |
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516 <P>Die Plugins können auch Schalter haben, welche ihr Verhalten |
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517 ändern. Diese Schalter werden im unteren Abschnitt im Detail beschrieben. |
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518 Ein Plugin wird mit den Standardeinstellungen gestartet, wenn dem Plugin |
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519 Optionen mitgegeben werden. Hier ist ein Beispiel, wie man Plugins in |
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520 Kombination mit pluginspezifischen Schaltern verwendet:</P> |
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521 |
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522 <P> <CODE>mplayer media.avi -aop |
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523 list=resample,format:fout=44100:format=0x8</CODE></P> |
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524 |
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525 <P>würde die Ausgabefrequenz des Resample-Plugins auf 44100Hz and das |
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526 Ausgabeformat des Format-Plugins auf AFMT_U8 stellen.</P> |
4996 | 527 |
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528 <P>Momentan können Audio-Plugins nicht im MEncoder verwendet |
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529 werden.</P> |
4996 | 530 |
531 | |
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532 <H5><A NAME="resample">2.3.2.4.1. Up/Downsampling</A></H5> |
3707 | 533 |
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534 <P>MPlayer unterstützt vollständiges up/downsampling von |
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535 Audiostreams. Dieses Plugin kann verwendet werden, wenn man eine Soundkarte |
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536 mit fester Frequenz besitzt, oder wenn man eine alte Soundkarte am Hals hat, |
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537 welche nur max 44.1kHz verarbeitet. Ob die Verwendung dieses Plugins |
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538 benötigt wird oder nicht erkennt MPlayer <B>automatisch</B>. |
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539 Dieses Plugin hat einen Schalter, <CODE>fout</CODE>, welcher benutzt wird, um |
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540 die gewünschte Ausgabefrequenz zu setzen. Der Standard ist 48kHz und wird |
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541 in <Hz> angegeben.</P> |
3707 | 542 |
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543 <P>Verwendung:<BR> <CODE>mplayer medium.avi -aop |
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544 list=resample:fout=<erforderliche Frequenz in Hz, z.B. 44100></CODE></P> |
3707 | 545 |
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546 <P>Hinweis: Die Ausgabefrequenz sollte nicht vom Standard-Wert hochskaliert |
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547 werden. Die Skalierung führt dazu, dass der Audio- und Video-Stream in |
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548 Zeitlupe wiedergegeben werden, und man erhält zusätzlich |
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549 Audioverzerrungen.</P> |
3707 | 550 |
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551 |
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552 <H5><A NAME="surround_decoding">2.3.2.4.2. Surround-Sound-Decodierung</A></H5> |
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553 |
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554 <P>MPlayer hat ein Audio-Plugin das matrix-codierten Surround Sound |
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555 decodieren kann. Dolby Surround ist ein Beispiel eines matrix-codierten |
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556 Formats. Viele Dateien mit zwei Audiokanälen enthalten tatsächlich |
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557 matrix Surround Sound. Um dieses Feature zu benutzen, brauchst du eine |
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558 Soundkarte mit mindestens vier Kanälen.</P> |
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559 |
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560 <P>Verwendung:<BR> |
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561 <CODE>mplayer medium.avi -aop list=surround</CODE></P> |
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562 |
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563 |
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564 <H5><A NAME="format">2.3.2.4.3. Sample-Format-Konvertierer</A></H5> |
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565 |
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566 <P>Wenn deine Soundkarte keinen signed 16bit <CODE>int</CODE> Datentyp |
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567 unterstützt, kann dieses Plugin benutzt werden, um das Format in eines zu |
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568 ändern, das deine Soundkarte versteht. Es hat nur einen Schalter |
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569 <CODE>format</CODE>, welcher auf eine Nummer gestellt werden kann, die man der |
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570 Datei <CODE>libao2/afmt.h</CODE> findet. Dieses Plugin wird praktisch nie |
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571 gebraucht und ist für erfahrene Anwender gedacht. Bedenke, dass dieses |
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572 Plugin nur das Sampleformat und nicht die Samplefrequenz oder die Anzahl |
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573 Kanäle ändert.</P> |
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574 |
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575 <P>Verwendung:<BR> |
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576 <CODE>mplayer medium.avi -aop |
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577 list=format:format=<required output format></CODE></P> |
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578 |
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579 |
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580 <H5><A NAME="delay">2.3.2.4.4. Verzögerung</A></H5> |
4996 | 581 |
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582 <P>Dieses Plugin verzögert den Ton und ist als Beispiel gedacht, wie man |
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583 neue Plugins schreibt. Es kann für nichts sinnvolles aus der |
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584 Anwenderperspektive verwendet werden und ist hier nur aus Gründen der |
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585 Vollständigkeit erwähnt. Verwende dieses Plugin nicht, wenn du kein |
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586 Entwickler bist.</P> |
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587 |
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588 |
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589 <H5><A NAME="volume">2.3.2.4.5. Software Lautstärkeregelung</A></H5> |
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591 <P>Dieses Plugin ist ein Software-Ersatz für die Lautstärkereglung |
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592 und kann in Maschinen mit kaputtem Mixer-Device verwendet werden. Es kann auch |
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593 verwendet werden, um die Ausgabelautstärke von MPlayer zu |
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594 ändern, ohne die Einstellung der PCM-Lautstärke im Mixer zu |
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595 ändern. Es gibt einen Schalter <CODE>volume</CODE>, welcher verwendet |
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596 wird, um den anfänglichen Schallpegel festzulegen. Der anfängliche |
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597 Schallpegel kann auf Werte zwischen 0 und 255 festgelegt werden (normal auf |
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598 101, was 0db Verstärkung entspricht). Benutze dieses Plugin mit Vorsicht, |
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599 da es den Signal/Rauschabstand verringern kann.In den meisten Fällen ist |
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600 es das beste, den Reglere für PCM auf das Maximum zu stellen, |
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601 dieses Plugin nicht zu verwenden und die Lautstärke zu deinen Boxen mit |
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602 der Master-Lautstärkeregelung zu kontrollieren. Wenn ein externer |
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603 Verstärker mit dem Computer verbunden ist (was fast immer der Fall ist), |
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604 kann der Geräuschepegel durch die Anpassung des Master-Stufe und der |
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605 Lautstärkeknöpfe des Verstärkers minimiert werden, bis das |
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606 Hintergrundrauschen verschwindet.</P> |
4996 | 607 |
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608 <P>Verwendung:<BR> |
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609 <CODE>mplayer media.avi -aop |
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610 list=volume:volume=<0-255></CODE></P> |
4996 | 611 |
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612 <P>Dieses Plugin besitzt auch einen Kompressor oder "soft-clipping"- |
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613 Fähigkeiten. Die Kompression kann benutzt werden, wenn der Dynamikbereich |
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614 des Sounds sehr hoch ist, oder wenn der Dynamikbereich der Lautsprecher sehr |
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615 niedrig ist. Sei dir bewusst, dass dieses Feature Verzerrungen verursacht. Es |
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616 sollte erst als letzte Möglichkeit in Betracht gezogen werden.</P> |
5185 | 617 |
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618 <P>Verwendung:<BR> |
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619 <CODE>mplayer media.avi -aop |
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620 list=volume:softclip</CODE></P> |
5185 | 621 |
4996 | 622 |
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623 <H5><A NAME="extrastereo">2.3.2.4.6. Extrastereo</A></H5> |
4996 | 624 |
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625 <P>Dieses Plugin erhöht linear die Differenz zwischen dem linken und |
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626 rechten Kanal (wie das XMMS Extrastereo-Plugin), was zum Teil zu "live"- |
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627 Effekten bei der Wiedergabe führt.</P> |
4996 | 628 |
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629 <P>Verwendung:<BR> |
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630 <CODE>mplayer media.avi -aop list=extrastereo</CODE><BR> |
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631 <CODE>mplayer media.avi -aop list=extrastereo:mul=3.45</CODE></P> |
4996 | 632 |
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633 <P>Der Koeffizient (<CODE>mul</CODE>) ist eine Fließkommazahl, |
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634 welche standardmäßig auf 2.5 gestellt ist. Wenn man den Wert auf |
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635 0.0 setzt, erhält man einen Mono-Ton (Durchschnitt beider Kanäle). |
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636 Wenn man den Wert auf 1.0 setzt, bleibt der Ton gleich. Wenn man den Wert auf |
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637 -1.0 setzt, werden der linke und rechte Kanal vertauscht.</P> |
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638 |
5185 | 639 |
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640 <H5><A NAME="normalizer">2.3.2.4.7. Lautstärkenormalisierer</A></H5> |
5185 | 641 |
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642 <P>Dieses Plugin maximiert die Lautstärke, ohne den Ton zu |
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643 verzerren.</P> |
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644 |
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645 <P>Verwendung:<BR> |
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646 <CODE>mplayer media.avi -aop list=volnorm</CODE></P> |
5185 | 647 |
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648 |
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649 </BODY> |
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650 </HTML> |