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author | diego <diego@b3059339-0415-0410-9bf9-f77b7e298cf2> | 2005-02-24 11:00:45 +0000 |
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committer | diego <diego@b3059339-0415-0410-9bf9-f77b7e298cf2> | 2005-02-24 11:00:45 +0000 |
commit | 201691be524b294e30ff5a935e625655bdf38555 (patch) | |
tree | e99530c09156cc865c89a0d258c9b47aa98e411f /DOCS/xml | |
parent | 55aa474d7e70016da5f36edd2251de1b476a50ba (diff) |
Move audio filter descriptions to the man page.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@14788 b3059339-0415-0410-9bf9-f77b7e298cf2
Diffstat (limited to 'DOCS/xml')
-rw-r--r-- | DOCS/xml/en/audio.xml | 699 |
1 files changed, 0 insertions, 699 deletions
diff --git a/DOCS/xml/en/audio.xml b/DOCS/xml/en/audio.xml index 2dab6d4bb8..2edc83cff0 100644 --- a/DOCS/xml/en/audio.xml +++ b/DOCS/xml/en/audio.xml @@ -55,703 +55,4 @@ DEMUXER: Too many (945 in 8390980 bytes) video packets in the buffer!<!-- </para> </sect2> - -</sect1> - -<sect1 id="af"> -<title>Audio filters</title> -<para> - Audio filters allow changing the properties of the audio data before the - sound reaches the sound card. The activation and deactivation of the filters - is normally automated but can be overridden. The filters are activated when - the properties of the audio data differ from those required by the sound card - and deactivated if unnecessary. The <option>-af filter1,filter2,...</option> - option is used to override the automatic activation of filters or to insert - filters that are not automatically inserted. The filters will be executed as - they appear in the comma separated list. -</para> - -<para> -Example: -<screen>mplayer -af resample,pan <replaceable>movie.avi</replaceable></screen> -would run the sound through the resampling filter followed by the pan filter. -Observe that the list must not contain any spaces, else it will fail. -</para> - -<para> -The filters often have options that change their behavior. These options -are explained in detail in the sections below. A filter will execute using -default settings if its options are omitted. Here is an example of how to use -filters in combination with filter specific options: -<screen>mplayer -af resample=11025,pan=1:0.5:0.5 -channels 1 -srate 11025 <replaceable>media.avi</replaceable></screen> -would set the output frequency of the resample filter to 11025Hz and downmix -the audio to 1 channel using the pan filter. -</para> - - -<para> -The overall execution of the filter layer is controlled using the -<option>-af-adv</option> option. This option has two suboptions: -</para> - -<para> -<option>force</option> is a bit field that controls how the filters -are inserted and what speed/accuracy optimizations they use: -</para> - -<variablelist> -<varlistentry> -<term><option>0</option></term> -<listitem><para> -Use automatic insertion of filters and optimize according to CPU speed. -</para></listitem> -</varlistentry> - -<varlistentry> -<term><option>1</option></term> -<listitem><para> -Use automatic insertion of filters and optimize for the highest speed. -<emphasis>Warning:</emphasis> Some features in the audio filters may -silently fail, and the sound quality may drop. -</para></listitem> -</varlistentry> - -<varlistentry> -<term><option>2</option></term> -<listitem><para> -Use automatic insertion of filters and optimize for quality. -</para></listitem> -</varlistentry> - -<varlistentry> -<term><option>3</option></term> -<listitem><para> -Use no automatic insertion of filters and no optimization. -<emphasis>Warning:</emphasis> It may be possible to crash <application>MPlayer</application> -using this setting. -</para></listitem> -</varlistentry> - -<varlistentry> -<term><option>4</option></term> -<listitem><para> -Use automatic insertion of filters according to 0 above, -but use floating point processing when possible. -</para></listitem> -</varlistentry> - -<varlistentry> -<term><option>5</option></term> -<listitem><para> -Use automatic insertion of filters according to 1 above, -but use floating point processing when possible. -</para></listitem> -</varlistentry> - -<varlistentry> -<term><option>6</option></term> -<listitem><para> -Use automatic insertion of filters according to 2 above, -but use floating point processing when possible. -</para></listitem> -</varlistentry> - -<varlistentry> -<term><option>7</option></term> -<listitem><para> -Use no automatic insertion of filters according to 3 above, -and use floating point processing when possible. -</para></listitem> -</varlistentry> -</variablelist> - -<para> -<option>list</option> is an alias for the -af option. -</para> - -<para> -The filter layer is also affected by the following generic options: -</para> - -<variablelist> -<varlistentry> -<term><option>-v</option></term> -<listitem><para> -Increases the verbosity level and makes most filters print out extra -status messages. -</para></listitem> -</varlistentry> - -<varlistentry> -<term><option>-channels</option></term> -<listitem><para> -This option sets the number of output channels you would like your -sound card to use. It also affects the number of channels that are -being decoded from the media. If the media contains less channels -than requested the channels filter (see below) will automatically -be inserted. The routing will be the default routing for the channels -filter. -</para></listitem> -</varlistentry> - -<varlistentry> -<term><option>-srate</option></term> -<listitem><para> -This option selects the sample rate you would like your sound card -to use (of course the cards have limits on this). If the sample frequency -of your sound card is different from that of the current media, the resample -filter (see below) will be inserted into the audio filter layer to compensate -for the difference. -</para></listitem> -</varlistentry> -<varlistentry> -<term><option>-format</option></term> -<listitem><para> -This option sets the sample format between the audio filter layer and the -sound card. If the requested sample format of your sound card is different -from that of the current media, a format filter (see below) will be inserted -to rectify the difference. -</para></listitem> -</varlistentry> -</variablelist> - -<sect2 id="af_resample"> -<title>Up/Downsampling</title> - -<para> -<application>MPlayer</application> fully supports sound up/down-sampling through the -<option>resample</option> filter. It can be used if you -have a fixed frequency sound card or if you are stuck with an old sound card -that is only capable of max 44.1kHz. This filter is automatically enabled if -it is necessary, but it can also be explicitly enabled on the command line. It -has three options: -</para> - -<variablelist> -<varlistentry> -<term><option>srate <8000-192000></option></term> -<listitem><para> - is an integer used for setting the output sample - frequency in Hz. The valid range for this parameter is 8kHz to 192kHz. If - the input and output sample frequency are the same or if this parameter is - omitted the filter is automatically unloaded. A high sample frequency - normally improves the audio quality, especially when used in combination - with other filters. -</para></listitem> -</varlistentry> - -<varlistentry> -<term><option>sloppy</option></term> -<listitem><para> - is an optional binary parameter that allows the output frequency to differ - slightly from the frequency given by <option>srate</option>. This option - can be used if the startup of the playback is extremely slow. It is enabled - by default. -</para></listitem> -</varlistentry> - -<varlistentry> -<term><option>type <0-2></option></term> -<listitem><para> - is an optional integer between <literal>0</literal> and <literal>2</literal> that - selects which resampling method to use. Here <literal>0</literal> represents - linear interpolation as resampling method, <literal>1</literal> represents - resampling using a poly-phase filter-bank and integer processing and - <literal>2</literal> represents resampling using a poly-phase filter-bank and - floating point processing. Linear interpolation is extremely fast, but - suffers from poor sound quality especially when used for up-sampling. The - best quality is given by <literal>2</literal> but this method also suffers from - the highest CPU load. -</para></listitem> -</varlistentry> -</variablelist> - -<para>Example: -<screen>mplayer -af resample=44100:0:0</screen> -would set the output frequency of the resample filter to 44100Hz using exact output -frequency scaling and linear interpolation. -</para> -</sect2> - -<sect2 id="af_channels"> -<title>Changing the number of channels</title> -<para> -The <option>channels</option> filter can be used for adding and removing -channels, it can also be used for routing or copying channels. It is -automatically enabled when the output from the audio filter layer differs from -the input layer or when it is requested by another filter. This filter unloads -itself if not needed. The number of options is dynamic: -</para> - -<variablelist> -<varlistentry> -<term><option>nch <1-6></option></term> -<listitem><para> - is an integer between <literal>1</literal> and <literal>6</literal> that is used - for setting the number of output channels. This option is required, leaving it - empty results in a runtime error. -</para></listitem> -</varlistentry> - -<varlistentry> -<term><option>nr <1-6></option></term> -<listitem><para> - is an integer between <literal>1</literal> and <literal>6</literal> that is used - for specifying the number of routes. This parameter is optional. If it is - omitted the default routing is used. -</para></listitem> -</varlistentry> - -<varlistentry> -<term><option>from1:to1:from2:to2:from3:to3...</option></term> -<listitem><para> - are pairs of numbers between <literal>0</literal> and <literal>5</literal> - that define where each channel should be routed. -</para></listitem> -</varlistentry> -</variablelist> - -<para> - If only <option>nch</option> is given the default routing is used, it works - as follows: If the number of output channels is bigger than the number of input - channels empty channels are inserted (except mixing from mono to stereo, then - the mono channel is repeated in both of the output channels). If the number of - output channels is smaller than the number of input channels the exceeding - channels are truncated. -</para> - -<para> -Example 1: -<screen>mplayer -af channels=4:4:0:1:1:0:2:2:3:3 <replaceable>media.avi</replaceable></screen> -would change the number of channels to 4 and set up 4 routes that swap -channel 0 and channel 1 and leave channel 2 and 3 intact. Observe that -if media containing two channels was played back, channels 2 and 3 would -contain silence but 0 and 1 would still be swapped. -</para> - -<para> -Example 2: -<screen>mplayer -af channels=6:4:0:0:0:1:0:2:0:3 <replaceable>media.avi</replaceable></screen> -would change the number of channels to 6 and set up 4 routes that copy -channel 0 to channels 0 to 3. Channel 4 and 5 will contain silence. -</para> -</sect2> - -<sect2 id="af_format"> -<title>Sample format converter</title> -<para> -The <option>format</option> filter converts between different sample formats. It - is automatically enabled when needed by the sound card or another filter. -</para> - -<variablelist> -<varlistentry> -<term><option>bps <number></option></term> -<listitem><para> - can be <literal>1</literal>, <literal>2</literal> or <literal>4</literal> and - denotes the number of bytes per sample. This option is required, leaving it empty - results in a runtime error. -</para></listitem> -</varlistentry> - -<varlistentry> -<term><option>f <format></option></term> -<listitem><para> - is a text string describing the sample format. The string is a - concatenated mix of: <option>alaw</option>, <option>mulaw</option> or - <option>imaadpcm</option>, <option>float</option> or <option>int</option>, - <option>unsigned</option> or <option>signed</option>, <option>le</option> or - <option>be</option> (little- or big-endian). This option is required, - leaving it empty results in a runtime error. -</para></listitem> -</varlistentry> -</variablelist> - -<para> -Example: -<screen>mplayer -af format=4:float <replaceable>media.avi</replaceable></screen> -would set the output format to 4 bytes per sample floating point data. -</para> -</sect2> - -<sect2 id="af_delay"> -<title>Delay</title> -<para> -The <option>delay</option> filter delays the sound to the loudspeakers such that -the sound from the different channels arrives at the listening position -simultaneously. -It is only useful if you have more than 2 loudspeakers. This filter has a -variable number of parameters: -</para> - -<variablelist> -<varlistentry> -<term><option>d1:d2:d3...</option></term> -<listitem><para> - are floating point numbers representing the delays in ms that should be - imposed on the different channels. The minimum delay is 0ms and the maximum - is 1000ms. -</para></listitem> -</varlistentry> -</variablelist> - -<para> -To calculate the required delay for the different channels do as follows: -</para> - -<orderedlist> -<listitem><para> - Measure the distance to the loudspeakers in meters in relation to your - listening position, giving you the distances s1 to s5 (for a 5.1 system). - There is no point in compensating for the sub-woofer (you will not hear the - difference anyway). -</para></listitem> -<listitem><para> -Subtract the distances s1 to s5 from the maximum distance i.e. - s[i] = max(s) - s[i]; i = 1...5 -</para></listitem> -<listitem><para> -Calculate the required delays in ms as - d[i] = 1000*s[i]/342; i = 1...5 -</para></listitem> -</orderedlist> - -<para> -Example: -<screen>mplayer -af delay=10.5:10.5:0:0:7:0 <replaceable>media.avi</replaceable></screen> -would delay front left and right by 10.5ms, the two rear channels and the sub -by 0ms and the center channel by 7ms. -</para> - -</sect2> - -<sect2 id="af_volume"> -<title>Software volume control</title> -<para>Software volume control is implemented by the <option>volume</option> -audio filter. Use this filter with caution since it can reduce the signal to -noise ratio of the sound. In most cases it is best to set the level for the -PCM sound to max, leave this filter out and control the output level to your -speakers with the master volume control of the mixer. In case your sound card -has a digital PCM mixer instead of an analog one, and you hear distortion, -use the MASTER mixer instead. If there is an external amplifier connected to -the computer (this is almost always the case), the noise level can be minimized -by adjusting the master level and the volume knob on the amplifier until the -hissing noise in the background is gone. This filter has two options: -</para> - -<variablelist> -<varlistentry> -<term><option>v <-200 - +60></option></term> -<listitem><para> - is a floating point number between <literal>-200</literal> and <literal>+60</literal> - which represents the volume level in dB. The default level is 0dB. -</para></listitem> -</varlistentry> - -<varlistentry> -<term><option>c</option></term> -<listitem><para> - is a binary control that turns soft clipping on and off. Soft-clipping can - make the sound more smooth if very high volume levels are used. Enable this - option if the dynamic range of the loudspeakers is very low. Be aware that - this feature creates distortion and should be considered a last resort. -</para></listitem> -</varlistentry> -</variablelist> - -<para> -Example: -<screen>mplayer -af volume=10.1:0 <replaceable>media.avi</replaceable></screen> -would amplify the sound by 10.1dB and hard-clip if the sound level is too high. -</para> - -<para> -This filter has a second feature: It measures the overall maximum sound level -and prints out that level when <application>MPlayer</application> exits. -This volume estimate can be used for setting the sound level in -<application>MEncoder</application> such that the maximum dynamic range is utilized. -</para> -</sect2> - -<sect2 id="af_equalizer"> -<title>Equalizer</title> -<para> -The <option>equalizer</option> filter represents a 10 octave band graphic -equalizer, implemented using 10 IIR band pass filters. This means that -it works regardless of what type of audio is being played back. The center -frequencies for the 10 bands are: -</para> - -<informaltable> -<tgroup cols="2"> -<thead> - <row> - <entry>Band No.</entry><entry>Center frequency</entry> - </row> -</thead> -<tbody> - <row><entry>0</entry><entry>31.25 Hz</entry></row> - <row><entry>1</entry><entry>62.50 Hz</entry></row> - <row><entry>2</entry><entry>125.0 Hz</entry></row> - <row><entry>3</entry><entry>250.0 Hz</entry></row> - <row><entry>4</entry><entry>500.0 Hz</entry></row> - <row><entry>5</entry><entry>1.000 kHz</entry></row> - <row><entry>6</entry><entry>2.000 kHz</entry></row> - <row><entry>7</entry><entry>4.000 kHz</entry></row> - <row><entry>8</entry><entry>8.000 kHz</entry></row> - <row><entry>9</entry><entry>16.00 kHz</entry></row> -</tbody> -</tgroup> -</informaltable> - -<para> -If the sample rate of the sound being played back is lower than the center -frequency for a frequency band, then that band will be disabled. A known -bug with this filter is that the characteristics for the uppermost band -are not completely symmetric if the sample rate is close to the center -frequency of that band. This problem can be worked around by up-sampling -the sound using the resample filter before it reaches this filter. -</para> - -<para> -This filter has 10 parameters: -</para> - -<variablelist> -<varlistentry> -<term><option>g1:g2:g3...g10</option></term> -<listitem><para> -are floating point numbers between <literal>-12</literal> and <literal>+12</literal> -representing the gain in dB for each frequency band. -</para></listitem> -</varlistentry> -</variablelist> - -<para> -Example: -<screen>mplayer -af equalizer=11:11:10:5:0:-12:0:5:12:12 <replaceable>media.avi</replaceable></screen> -would amplify the sound in the upper and lower frequency region while -canceling it almost completely around 1kHz. -</para> -</sect2> - -<sect2 id="af_panning"> -<title>Panning filter</title> -<para> -Use the <option>pan</option> filter to mix channels arbitrarily. It is -basically a combination of the volume control and the channels filter. -There are two major uses for this filter: -</para> - -<orderedlist> -<listitem><para> -Down-mixing many channels to only a few, stereo to mono for example. -</para></listitem> -<listitem><para> -Varying the "width" of the center speaker in a surround sound system. -</para></listitem> -</orderedlist> - -<para> -This filter is hard to use, and will require some tinkering before the -desired result is obtained. The number of options for this filter -depends on the number of output channels: -</para> - -<variablelist> -<varlistentry> -<term><option>nch <1-6></option></term> -<listitem><para> -is an integer between <literal>1</literal> and <literal>6</literal> and is used -for setting the number of input channels. This option is required, leaving it -empty results in a runtime error. -</para></listitem> -</varlistentry> - -<varlistentry> -<term><option>l00:l01:l02:..l10:l11:l12:...ln0:ln1:ln2:...</option></term> -<listitem><para> -are floating point values between <literal>0</literal> and <literal>1</literal>. -<option>l[i][j]</option> determines how much of input channel j is mixed into -output channel i. -</para></listitem> -</varlistentry> -</variablelist> - -<para> -Example 1: -<screen>mplayer -af pan=1:0.5:0.5 -channels 1 <replaceable>media.avi</replaceable></screen> -would down-mix from stereo to mono. -</para> - -<para> -Example 2: -<screen>mplayer -af pan=3:1:0:1:0.5:0.5 -channels 3 <replaceable>media.avi</replaceable></screen> -would give 3 channel output leaving channels 0 and 1 intact, and mix -channels 0 and 1 into output channel 2 (which could be sent to a -sub-woofer for example). -</para> -</sect2> - -<sect2 id="af_sub"> -<title>Sub-woofer</title> -<para> -The <option>sub</option> filter adds a sub woofer channel to the audio -stream. The audio data used for creating the sub-woofer channel is an -average of the sound in channel 0 and channel 1. The resulting sound is -then low-pass filtered by a 4th order Butterworth filter with a default -cutoff frequency of 60Hz and added to a separate channel in the audio -stream. Warning: Disable this filter when you are playing DVDs with Dolby -Digital 5.1 sound, otherwise this filter will disrupt the sound to the -sub-woofer. This filter has two parameters: -</para> - -<variablelist> -<varlistentry> -<term><option>fc <20-300></option></term> -<listitem><para> - is an optional floating point number used for setting the cutoff frequency - for the filter in Hz. The valid range is 20Hz to 300Hz. For the best result - try setting the cutoff frequency as low as possible. This will improve the - stereo or surround sound experience. The default cutoff frequency is 60Hz. -</para></listitem> -</varlistentry> - -<varlistentry> -<term><option>ch <0-5></option></term> -<listitem><para> - is an optional integer between <literal>0</literal> and <literal>5</literal> - which determines the channel number in which to insert the sub-channel audio. - The default is channel number <literal>5</literal>. Observe that the number of - channels will automatically be increased to <replaceable>ch</replaceable> if - necessary. -</para></listitem> -</varlistentry> -</variablelist> - -<para> -Example: -<screen>mplayer -af sub=100:4 -channels 5 <replaceable>media.avi</replaceable></screen> -would add a sub-woofer channel with a cutoff frequency of -100Hz to output channel 4. -</para> -</sect2> - -<sect2 id="af_surround"> -<title>Surround-sound decoder</title> -<para> -Matrix encoded surround sound can be decoded by the <option>surround</option> -filter. Dolby Surround is an example of a matrix encoded format. Many files -with 2 channel audio actually contain matrixed surround sound. To use this -feature you need a sound card supporting at least 4 channels. This filter has -one parameter: -</para> - -<variablelist> -<varlistentry> -<term><option>d <0-1000></option></term> -<listitem><para> -is an optional floating point number between <literal>0</literal> and -<literal>1000</literal> used for setting the delay time in ms for the -rear speakers. This delay should be set as follows: if d1 is the distance -from the listening position to the front speakers and d2 is the distance -from the listening position to the rear speakers, then the delay d should -be set to 15ms if d1 <= d2 and to 15 + 5*(d1-d2) if d1 > d2. The default -value for d is 20ms. -</para></listitem> -</varlistentry> -</variablelist> - -<para> -Example: -<screen>mplayer -af surround=15 -channels 4 <replaceable>media.avi</replaceable></screen> -would add surround sound decoding with 15ms delay for the sound to the -rear speakers. -</para> -</sect2> - -<sect2 id="af_export"> -<title>Audio Exporter</title> -<para> -This audio filter exports the incoming signal to other processes using memory -mapping (<literal>mmap()</literal>). Memory mapped areas contain a header: - -<programlisting> -int nch /*number of channels*/ -int size /*buffer size*/ -unsigned long long counter /*Used to keep sync, it's updated - every time new data is exported.*/ -</programlisting> - -The rest is payload (non-interleaved) 16bit data. -</para> - -<variablelist> -<varlistentry> -<term><option>mmapped_file</option></term> -<listitem><para> -The file you want this filter to export to. The default is to map to -<filename>~/.mplayer/mplayer-af_export</filename>. -</para></listitem> -</varlistentry> - -<varlistentry> -<term><option>nsamples</option></term> -<listitem><para> -Number of samples per channel. The default is <literal>512</literal> samples. -</para></listitem> -</varlistentry> -</variablelist> - -<para> -Example: -<screen>mplayer -af export=/tmp/mplayer-af_export:1024 <replaceable>media.avi</replaceable></screen> -would export 1024 samples per channel to <filename>/tmp/mplayer-af_export</filename>. -</para> -</sect2> - -<sect2 id="af_extrastereo"> -<title>Extrastereo</title> - -<para> -This audio filter (linearly) increases the difference between left and -right channels (like the <application>XMMS</application> extrastereo -plugin) which adds some sort of "live" effect to playback. -This filter has one parameter:</para> - -<variablelist> -<varlistentry> -<term><option>mul</option></term> -<listitem><para> -is the difference coefficient, an optional floating point number that defaults -to <literal>2.5</literal>. If you set it to <literal>0.0</literal>, you will -have mono sound (average of both channels). If you set it to -<literal>1.0</literal>, sound will be unchanged, if you set it to -<literal>-1.0</literal>, left and right channels will be swapped. -</para></listitem> -</varlistentry> -</variablelist> - -<para> -Usage: -<screen> -mplayer -af extrastereo <replaceable>media.avi</replaceable> -mplayer -af extrastereo=3.45 <replaceable>media.avi</replaceable><!-- ---></screen> -</para> -</sect2> - -<sect2 id="af_volnorm"> -<title>Volume normalizer</title> - -<para> -This audio filter maximizes the volume without distorting the sound. -</para> - -<para> -Usage: -<screen>mplayer -af volnorm <replaceable>media.avi</replaceable></screen> -</para> -</sect2> - </sect1> |