Mercurial > mplayer.hg
view DOCS/xml/en/audio.xml @ 14731:0cf5722252f8
Point to local_changes.diff.
author | diego |
---|---|
date | Sat, 19 Feb 2005 02:19:21 +0000 |
parents | 5eb0f60a6962 |
children | 346ace66cdb4 |
line wrap: on
line source
<?xml version="1.0" encoding="iso-8859-1"?> <!-- $Revision$ --> <sect1 id="audio"> <title>Audio output devices</title> <sect2 id="sync"> <title>Audio/Video synchronization</title> <para> Linux sound card drivers have compatibility problems. This is because <application>MPlayer</application> relies on an in-built feature of <emphasis>properly</emphasis> coded sound drivers that enable them to maintain correct audio/video sync. Regrettably, some driver authors don't take the care to code this feature since it is not needed for playing MP3s or sound effects. </para> <para> Other media players like <ulink url="http://avifile.sf.net">aviplay</ulink> or <ulink url="http://xinehq.de">xine</ulink> possibly work out-of-the-box with these drivers because they use "simple" methods with internal timing. Measuring showed that their methods are not as efficient as <application>MPlayer</application>'s. </para> <para> Using <application>MPlayer</application> with a properly written audio driver will never result in A/V desyncs related to the audio, except only with very badly created files (check the man page for workarounds). </para> <para> If you happen to have a bad audio driver, try the <option>-autosync</option> option, it should sort out your problems. See the man page for detailed information. </para> </sect2> <sect2 id="troubleshooting"> <title>Troubleshooting</title> <para>Some notes:</para> <para> If you have ALSA version 0.5, then you almost always have to use <option>-ao alsa5</option>, since ALSA 0.5 has buggy OSS emulation code, and will <emphasis role="bold">crash <application>MPlayer</application></emphasis> with a message like this: <screen> DEMUXER: Too many (945 in 8390980 bytes) video packets in the buffer!<!-- --></screen> </para> <para> If the sound clicks when playing from CD-ROM, turn on IRQ unmasking as described in the <link linkend="drives">CD-ROM</link> section. </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>