Mercurial > mplayer.hg
changeset 8750:f6323ff433aa
New audio filter documentation by Anders Johannsson with some structural
modifications by myself.
| author | diego |
|---|---|
| date | Fri, 03 Jan 2003 22:29:16 +0000 |
| parents | 8d29bc9a5836 |
| children | f117a4150786 |
| files | DOCS/documentation.html DOCS/sound.html |
| diffstat | 2 files changed, 365 insertions(+), 35 deletions(-) [+] |
line wrap: on
line diff
--- a/DOCS/documentation.html Fri Jan 03 21:54:54 2003 +0000 +++ b/DOCS/documentation.html Fri Jan 03 22:29:16 2003 +0000 @@ -185,17 +185,28 @@ </UL> <LI><A HREF="sound.html">2.3.2 Audio output devices</A> <UL> - <LI><A HREF="sound.html#sync">2.3.2.1 Description of MPlayer's A/V sync method</A></LI> + <LI><A HREF="sound.html#sync">2.3.2.1 Audio/Video synchronisation</A></LI> <LI><A HREF="sound.html#experiences">2.3.2.2 Sound card experiences, recommendations</A></LI> - <LI><A HREF="sound.html#plugins">2.3.2.3 Audio plugins</A> + <LI><A HREF="sound.html#af">2.3.2.3 Audio filters</A> <UL> - <LI><A HREF="sound.html#resample">2.3.2.3.1 Up/Downsampling</A></LI> - <LI><A HREF="sound.html#surround_decoding">2.3.2.3.2 Surround Sound decoding</A></LI> - <LI><A HREF="sound.html#format">2.3.2.3.3 Sample format converter</A></LI> - <LI><A HREF="sound.html#delay">2.3.2.3.4 Delay</A></LI> - <LI><A HREF="sound.html#volume">2.3.2.3.5 Software volume control</A></LI> - <LI><A HREF="sound.html#extrastereo">2.3.2.3.6 Extrastereo</A></LI> - <LI><A HREF="sound.html#normalizer">2.3.2.3.7 Volume Normalizer</A></LI> + <LI><A HREF="sound.html#af_resample">2.3.2.3.1 Up/Downsampling</A></LI> + <LI><A HREF="sound.html#af_channels">2.3.2.3.2 Changing the number of channels</A></LI> + <LI><A HREF="sound.html#af_format">2.3.2.3.3 Sample format converter</A></LI> + <LI><A HREF="sound.html#af_delay">2.3.2.3.4 Delay</A></LI> + <LI><A HREF="sound.html#af_volume">2.3.2.3.5 Software volume control</A></LI> + <LI><A HREF="sound.html#af_equalizer">2.3.2.3.6 Equalizer</A></LI> + <LI><A HREF="sound.html#af_panning">2.3.2.3.7 Panning filter</A></LI> + </UL> + </LI> + <LI><A HREF="sound.html#plugins">2.3.2.4 Audio plugins (deprecated)</A> + <UL> + <LI><A HREF="sound.html#resample">2.3.2.4.1 Up/Downsampling</A></LI> + <LI><A HREF="sound.html#surround_decoding">2.3.2.4.2 Surround Sound decoding</A></LI> + <LI><A HREF="sound.html#format">2.3.2.4.3 Sample format converter</A></LI> + <LI><A HREF="sound.html#delay">2.3.2.4.4 Delay</A></LI> + <LI><A HREF="sound.html#volume">2.3.2.4.5 Software volume control</A></LI> + <LI><A HREF="sound.html#extrastereo">2.3.2.4.6 Extrastereo</A></LI> + <LI><A HREF="sound.html#normalizer">2.3.2.4.7 Volume Normalizer</A></LI> </UL> </LI> </UL>
--- a/DOCS/sound.html Fri Jan 03 21:54:54 2003 +0000 +++ b/DOCS/sound.html Fri Jan 03 22:29:16 2003 +0000 @@ -12,12 +12,12 @@ <H3><A NAME="audio">2.3.2 Audio output devices</A></H3> -<H4><A NAME="sync">2.3.2.1 Description of MPlayer's A/V sync method</A></H4> +<H4><A NAME="sync">2.3.2.1 Audio/Video synchronisation</A></H4> <P>MPlayer's audio interface is called <I>libao2</I>. It currently contains these drivers:</P> -<TABLE BORDER=0> +<TABLE BORDER="0"> <TR><TD COLSPAN=4><P><B>General:</B></P></TD></TR> <TR><TD> </TD><TD VALIGN=top>oss</TD><TD> </TD><TD>OSS (ioctl) driver (supports hardware AC3 passthrough)</TD></TR> <TR><TD></TD><TD VALIGN=top>sdl</TD><TD></TD><TD>SDL driver (supports <B>ESD</B>, <B>ARTS</B> etc)</TD></TR> @@ -29,17 +29,17 @@ </TABLE> <P>Fact is, Linux sound card drivers have compatibility problems. The cause - is that MPlayer uses a feature of normally coded audio drivers to maintain - audio/video sync. Regrettably, some driver authors don't care of this - function: it isn't needed for playing MP3s, or sound effects.</P> + is that MPlayer uses a feature that well coded audio drivers implement to + maintain audio/video sync. Regrettably, some driver authors do not care about + this function, it is not needed for playing MP3s or for sound effects.</P> <P>Other media players like aviplay or xine possibly work out-of-the-box with these drivers because they use "simple" methods with internal timing. A note: time showed their methods aren't AS efficient as MPlayer's.</P> -<P>Using MPlayer with a correctly written audio driver won't ever give you A/V - desyncs related to the audio, only with very badly created files (check the - documentation for workarounds!).</P> +<P>With a correctly written audio driver MPlayer will never create audio related + A/V desynchronisation, unless your file is badly broken. Some options to work + around these problems are described in the man page).</P> <P>If you happen to have a bad audio driver, try the <CODE>-autosync</CODE> option, it should sort out your problems. See the man page for detailed @@ -50,9 +50,9 @@ <UL> <LI>If you have an OSS driver, first try <CODE>-ao oss</CODE> (this is the default). If you experience glitches, halts or anything out of the - ordinary, try <CODE>-ao sdl</CODE> (NOTE: you need to have SDL libraries + ordinary, try <CODE>-ao sdl</CODE> (NOTE: You need to have SDL libraries and header files installed). The SDL audio driver helps in a lot of cases - and also supports ESD, ARTS. (ESD is the sound daemon + and also supports ESD and ARTS. (ESD is the sound daemon from GNOME, ARTS is from KDE.)</LI> <LI>If you have ALSA version 0.5, then you almost always have to use <CODE>-ao alsa5</CODE> , since ALSA 0.5 has buggy OSS emulation code, and @@ -66,9 +66,10 @@ <H4><A NAME="experiences">2.3.2.2 Sound Card experiences, recommendations</A></H4> -<TABLE BORDER=0 WIDTH="100%"> +<TABLE BORDER="0" WIDTH="100%"> <TR><TD COLSPAN=3><B>VIA onboard chipset (via82cxxx) 48kHz only</B></TD></TR> - <TR><TD></TD><TD>Driver:</TD><TD> from <A HREF="http://sourceforge.net/project/showfiles.php?group_id=3242&release_id=59602">sourceforge.net</A></TD></TR> + <TR><TD></TD><TD>Driver:</TD><TD> from the + <A HREF="http://sourceforge.net/project/showfiles.php?group_id=3242&release_id=59602">gkernel project</A></TD></TR> <TR><TD COLSPAN=3><B>Aureal Vortex 2</B></TD></TR> <TR><TD> </TD><TD>OSS:</TD><TD>no driver</TD></TR> @@ -135,10 +136,10 @@ <P>On Linux, a 2.4.x kernel is highly recommended. Kernel 2.2 is not tested.</P> -<P>If sound clicks when playing from CD-ROM, turn on IRQ unmasking, e.g. +<P>If the sound clicks when playing from CD-ROM, turn on IRQ unmasking, e.g. <CODE>hdparm -u1 /dev/cdrom</CODE> (<CODE>man hdparm</CODE>). This is - generally beneficial and described more detailed in the <A - HREF="cd-dvd.html#drives">CD-ROM section</A>.</P> + generally beneficial and described in more detail in the + <A HREF="cd-dvd.html#drives">CD-ROM section</A>.</P> <P>Sharing your sound card with another application like XMMS is <B>strongly discouraged</B>! If the other sound application is using ESD, start @@ -150,7 +151,325 @@ and your sound card(s) worked together.</P> -<H4><A NAME="plugins">2.3.2.3 Audio plugins</A></H4> +<H4><A NAME="af">2.3.2.3 Audio filters</A></H4> + +<P>The old audio plugins have been superseded by a new audio filter layer. Audio + filters are used for 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 <CODE>-af filter1,filter2,...</CODE> + switch 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.</P> + +<P>Example:<BR> + <CODE>mplayer -af resample,pan movie.avi </CODE></P> + +<P>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.</P> + +<P>The filters often have switches that change their behavior. These switches + are explained in detail in the sections below. A filter will execute using + default settings if its switches are omitted. Here is an example of how to use + filters in combination with filter specific switches:</P> + +<P> <CODE>mplayer -af resample=11025,pan=1:0.5:0.5 -channels 1 + -srate 11025 media.avi</CODE></P> + +<P>would set the output frequency of the resample filter to 11025Hz and downmix + the audio to 1 channel using the pan filter.</P> + +<P>Most filters respond to the <CODE>-v</CODE> switch, which makes the filters + print out status messages.</P> + +<P>The overall execution of the filter layer is controlled using the + <CODE>-af-adv</CODE> switch. This switch has two suboptions:</P> + +<DL> + <DT><CODE>force</CODE><DT> + <DD>is an integer between 0 and 3 that controls how the filters are inserted + and what speed/accuracy optimizations they use: + <DL> + <DT>0</DT> + <DD>Use automatic insertion of filters and optimize according to CPU + speed.</DD> + <DT>1</DT> + <DD>Use automatic insertion of filters and optimize for the highest speed. + If this option is set the processing of the audio data will be done + using fix point arithmetics. Warning: Some features in the audio filters + will silently fail, and the sound quality may drop.</DD> + <DT>2</DT> + <DD>Use automatic insertion of filters and optimize for quality. If this + option is set the processing of the audio data will be done using + floating point instructions and is therefore quite CPU intensive, but + gives a lot higher sound quality than fix point processing.</DD> + <DT>3</DT> + <DD>Use no automatic insertion of filters and no optimization. Warning: It + may be possible to crash MPlayer using this setting.</DD> + </DL> + + </DD> + + <DT><CODE>list</CODE></DT> + <DD>is an alias for the -af switch.</DD> +</DL> + + +<H5><A NAME="af_resample">2.3.2.3.1 Up/Down-sampling</A></H5> + +<P>MPlayer fully supports sound up/down-sampling. This filter 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 switches:</P> + +<DL> + <DT><CODE>srate</CODE></DT> + <DD>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.</DD> + + <DT><CODE>sloppy</CODE></DT> + <DD>is an optional binary parameter that allows the output frequency to differ + slightly from the frequency given by <CODE>srate</CODE>. This switch can be + used if the startup of the playback is extremely slow.</DD> + + <DT><CODE>fast</CODE><DT> + <DD>is an optional binary parameter that enables linear interpolation as + resampling method. Linear interpolation is extremely fast, but suffers from + poor sound quality especially when used for up-sampling.</DD> +</DL> + +<P>Example:<BR> + <CODE>mplayer -af resample=44100:0:1</CODE></P> + +<P>would set the output frequency of the resample filter to 44100Hz using exact + output frequency scaling and linear interpolation.</P> + + +<H5><A NAME="af_channels">2.3.2.3.2 Changing the number of channels</A></H5> + +<P>The <CODE>channels</CODE> 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 switches is dynamic:</P> + +<DL> + <DT><CODE>nch</CODE></DT> + <DD>is an integer between 1 and 6 that is used for setting the number of + output channels. This switch is required, leaving it empty results in a + runtime error.</DD> + + <DT><CODE>nr</CODE></DT> + <DD>is an integer between 1 and 6 that is used for specifying the number of + routes. This parameter is optional. If it is omitted the default routing is + used.</DD> + + <DT><CODE>from1:to1:from2:to2:from3:to3...</CODE></DT> + <DD>are pairs of numbers between 0 and 5 that define where each channel should + be routed.</DD> +</DL> + +<P>If only <CODE>nch</CODE> 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.</P> + +<P>Example 1:<BR> + <CODE>mplayer -af channels=4:4:0:1:1:0:2:2:3:3 media.avi </CODE></P> + +<P>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.</P> + +<P>Example 2:<BR> + <CODE>mplayer -af channels=6:4:0:0:0:1:0:2:0:3 media.avi </CODE></P> + +<P>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.</P> + + +<H5><A NAME="af_format">2.3.2.3.3 Sample format converter</A></H5> + +<P>This filter is a sample format converter. It is automatically enabled when + needed by the sound card or another filter.</P> + +<DL> + <DT><CODE>bps</CODE></DT> + <DD>can be 1, 2 or 4 and denotes the number of bytes per sample. This switch + is required, leaving it empty results in a runtime error.</DD> + + <DT><CODE>f</CODE></DT> + <DD>is a text string describing the sample format. The string is a + concatenated mix of: <CODE>alaw</CODE>, <CODE>mulaw</CODE> or + <CODE>imaadpcm</CODE>, <CODE>float</CODE> or <CODE>int</CODE>, + <CODE>unsigned</CODE> or <CODE>signed</CODE>, <CODE>le</CODE> or + <CODE>be</CODE> (little or big endian). This switch is required, leaving it + empty results in a runtime error.</DD> +</DL> + +<P>Example:<BR> + <CODE>mplayer media.avi -af format=4:float</CODE></P> + +<P>would set the output output format to 4 bytes per sample floating point + data.</P> + + +<H5><A NAME="af_delay">2.3.2.3.4 Delay</A></H5> + +<P>This filter delays the sound to the loudspeakers in order to make the sound + in the different channels arrive at the same time to the listening position. + It is only useful if you have more than 2 loudspeakers. This filter has a + variable number of parameters:</P> + +<DL> + <DT><CODE>d1:d2:d3...</CODE></DT> + <DD>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.</DD> +</DL> + +<P>To calculate the required delay for the different channels do as follows:</P> + +<OL> + <LI>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).</LI> + <LI>Subtract the distances s1 to s5 from the maximum distance i.e.<BR> + s[i] = max(s) - s[i]; i = 1...5</LI> + <LI>Calculated the required delays in ms as<BR> + d[i] = 1000*s[i]/342; i = 1...5 </LI> +</OL> + +<P>Example:<BR> + <CODE>mplayer -af delay=10.5:10.5:0:0:7:0 media.avi</CODE></P> + +<P>would delay front left and right by 10.5ms, the two rear channels and the sub + by 0ms and the center channel by 7ms.</P> + + +<H5><A NAME="af_volume">2.3.2.3.5 Software volume control</A></H5> + +<P>This filter is a software volume control. 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. + 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 switches:</P> + +<DL> + <DT><CODE>v</CODE></DT> + <DD>is a floating point number between -200 and +60 which represents the + volume level in dB. The default level is -10dB.</DD> + + <DT><CODE>c</CODE></DT> + <DD>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 + switch if the dynamic range of the loudspeakers is very low. Be aware that + this feature creates distortion and should be considered a last resort.</DD> +</DL> + +<P>Example:<BR> + <CODE>mplayer -af volume=10.1:0 media.avi</CODE></P> + +<P>would amplify the sound by 10.1dB and hard-clip if the sound level is too + high.</P> + +<P>This filter has a second feature: It measures the overall maximum sound level + and prints out that level when MPlayer exits. This volume estimate can be used + for setting the sound level in MEncoder such that the maximum dynamic range is + utilized.</P> + + +<H5><A NAME="af_equalizer">2.3.2.3.6 Equalizer</A></H5> + +<P> This filter is 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:</P> + +<TABLE BORDER="0" WIDTH="100%"> + <TR><TD>Band No.</TD><TD>Center frequency</TD></TR> + <TR><TD>0</TD><TD>31.25 Hz</TD></TR> + <TR><TD>1</TD><TD>62.50 Hz</TD></TR> + <TR><TD>2</TD><TD>125.0 Hz</TD></TR> + <TR><TD>3</TD><TD>250.0 Hz</TD></TR> + <TR><TD>4</TD><TD>500.0 Hz</TD></TR> + <TR><TD>5</TD><TD>1.000 kHz</TD></TR> + <TR><TD>6</TD><TD>2.000 kHz</TD></TR> + <TR><TD>7</TD><TD>4.000 kHz</TD></TR> + <TR><TD>8</TD><TD>8.000 kHz</TD></TR> + <TR><TD>9</TD><TD>16.00 kHz</TD></TR> +</TABLE> + +<P>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. </P> + +<P> This filter has 10 parameters:</P> + +<DL> + <DT><CODE>g1:g2:g3...g10</CODE></DT> + <DD>are floating point numbers between -12 to +12dB representing the gain in + dB for each frequency band.</DD> +</DL> + +<P>Example:<BR> + <CODE>mplayer -af equalizer=11:11:10:5:0:-12:0:5:12:12 media.avi</CODE></P> + +<P>would amplify the sound in the upper and lower frequency region while + canceling it almost completely around 1kHz.</P> + +<H5><A NAME="af_panning">2.3.2.3.7 Panning filter </A></H5> + +<P>This filter can be used for mixing the channels arbitrarily. It is basically + a combination of the volume control and the channels filter. There are two + major uses for this filter: </P> + +<OL> + <LI>Down-mixing many channels to only a few, stereo to mono for example.</LI> + <LI>Varying the "width" of the center speaker in a surround sound system.</LI> +</OL> + +<P>This filter is hard to use, and will require some tinkering before the + desired result is obtained. The number of switches for this filter depends on + the number of output channels:</P> + +<DL> + <DT><CODE>nch</CODE></DT> + <DD>is an integer between 1 and 6 and is used for setting the number of output + channels. This switch is required, leaving it empty results in a runtime + error.</DD> + + <DT><CODE>l00:l01:l02:..l10:l11:l12:...ln0:ln1:ln2:...</CODE></DT> + <DD>are floating point values between 0 and 1 that determine the level + <CODE>l[i][j]</CODE> that the input channel j is mixed into output channel + i.</DD> +</DL> + +<P>Example:<BR> + <CODE>mplayer -af pan=1:0.5:0.5 -channels 1 media.avi</CODE></P> + +<P>would down-mix from stereo to mono.</P> + + +<H2><STRONG>Note: Audio plugins have been deprecated by audio filters and will be + removed soon.</STRONG></H2> + +<H4><A NAME="plugins">2.3.2.4 Audio plugins (deprecated)</A></H4> <P>MPlayer has support for audio plugins. Audio plugins can be used for changing the properties of the audio data before the sound reaches the sound @@ -173,17 +492,17 @@ list=resample,format:fout=44100:format=0x8</CODE></P> <P>would set the output frequency of the resample plugin to 44100Hz and the - output format of the format plugin to AFMT_U8.</P> + output format of the format plugin to AFMT_U8.</P> <P>Currently audio plugins can not be used in MEncoder.</P> -<H5><A NAME="resample">2.3.2.3.1 Up/Downsampling</A></H5> +<H5><A NAME="resample">2.3.2.4.1 Up/Downsampling</A></H5> <P>MPlayer fully supports up/downsampling of the sound. This plugin 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. - Whether is usage of this plugin is neccessary or not, is <B>autodetected</B>. + Whether is usage of this plugin is necessary or not, is <B>autodetected</B>. This plugin has one switch: <CODE>fout</CODE> which is used for setting the desired output sample frequency. It defaults to 48kHz, and is given in @@ -198,7 +517,7 @@ in addition to audio distortion.</P> -<H5><A NAME="surround_decoding">2.3.2.3.2 Surround Sound decoding</A></H5> +<H5><A NAME="surround_decoding">2.3.2.4.2 Surround Sound decoding</A></H5> <P>MPlayer has an audio plugin that can decode matrix encoded surround sound. Dolby Surround is an example of a matrix encoded format. @@ -210,8 +529,8 @@ <H5><A NAME="format">2.3.2.3.3 Sample format converter</A></H5> - -<P>If your sound card driver does not support signed 16bit <CODE>int</CODE> data type, + +<P>If your sound card driver does not support signed 16bit <CODE>int</CODE> data type, this plugin can be used to change the format to one which your sound card can understand. It has one switch, <CODE>format</CODE>, which can be set to one of the numbers @@ -224,7 +543,7 @@ list=format:format=<required output format></CODE></P> -<H5><A NAME="delay">2.3.2.3.4 Delay</A></H5> +<H5><A NAME="delay">2.3.2.4.4 Delay</A></H5> <P>This plugin delays the sound and is intended as an example of how to develop new plugins. It can not be used for anything useful from a users perspective @@ -232,7 +551,7 @@ plugin unless you are a developer.</P> -<H5><A NAME="volume">2.3.2.3.5 Software volume control</A></H5> +<H5><A NAME="volume">2.3.2.4.5 Software volume control</A></H5> <P>This plugin is a software replacement for the volume control, and can be used on machines with a broken mixer device. It can also be @@ -265,7 +584,7 @@ list=volume:softclip</CODE></P> -<H5><A NAME="extrastereo">2.3.2.3.6 Extrastereo</A></H5> +<H5><A NAME="extrastereo">2.3.2.4.6 Extrastereo</A></H5> <P>This plugin (linearly) increases the difference between left and right channels (like the XMMS extrastereo plugin) which gives some sort of "live" @@ -281,7 +600,7 @@ -1.0, left and right channels will be swapped.</P> -<H5><A NAME="normalizer">2.3.2.3.7 Volume normalizer</A></H5> +<H5><A NAME="normalizer">2.3.2.4.7 Volume normalizer</A></H5> <P>This plugin maximizes the volume without distorting the sound.</P>