comparison DOCS/man/en/mplayer.1 @ 14787:346ace66cdb4

Move audio filter descriptions to the man page.

14786:e1bf5e07962f

To get a full list of available audio filters, see \-af help.
.PP
Available filters are:
.
.TP
.B resample[=srate[:sloppy][:type]]
Changes the sample rate of the audio stream to an integer srate in Hz.
It only supports the 16-bit little-endian format.
With MEncoder, you need to also use \-srate <srate>.
.
.TP
.B lavcresample[=srate[:length[:linear[:count[:cutoff]]]]]
Changes the sample rate of the audio stream to an integer srate in Hz.
It only supports the 16-bit little-endian format.
With MEncoder, you need to also use \-srate <srate>.
.PD 0
.RSs
.IPs <srate>
the output sample rate

.B hrtf\ \ \ 
Head-related transfer function: Converts multichannel audio to
2 channel output for headphones, preserving the spatiality of the sound.
.
.TP
.B channels[=nch]
Change the number of channels to <nch> output channels.
If the number of output channels is bigger than the number of input channels
empty channels are inserted (except when 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.
.
.TP
.B format[=format]
Change the current sample format.
.PD 0
.RSs
.IPs <format>
Sets the desired format.
The general form is 'sbe', where 's' denotes the sign (either 's' for signed
or 'u' for unsigned), 'b' denotes the number of bits per sample (16, 24 or 32)
and 'e' denotes the endianness ('le' means little-endian, 'be' big-endian
and 'ne' the endianness of the computer MPlayer is running on).
Valid values (amongst others) are: 's16le', 'u32be' and 'u24ne'.
Exceptions to this rule are: u8, s8, floatle, floatbe, floatne, mulaw, alaw,
mpeg2, ac3 and imaadpcm.
.RE
.PD 1
.
.TP
.B volume[=v:sc]
Select the output volume level.
This filter is not reentrant and can therefore only be enabled once for every
audio stream.
.PD 0
.RSs
.IPs <v>\ \ 
Sets the desired gain in dB for all channels in the stream
from -200dB to +60dB (where -200dB mutes the sound
completely and +60dB equals a gain of 1000).
.IPs <sc>\ 
Enable soft clipping.
.RE
.PD 1
.
.TP
.B pan[=n:l01:l02:...l10:l11:l12:...ln0:ln1:ln2:...]
Mixes channels arbitrarily, see DOCS/\:HTML/\:en/\:audio.html for details.
An example how to downmix a six-channel file to two channels with this
filter can be found in the examples section near the end of the man page.
.PD 0
.RSs
.IPs <n>\ \ 
number of input channels (1\-6)
.IPs <lij>
How much of input channel j is mixed into output channel i.
.RE
.PD 1
.
.TP
.B sub[=fc:ch]
Add subwoofer channel.
.PD 0
.RSs
.IPs <fc>\ 
cutoff frequency for low-pass filter (20Hz to 300Hz) (default: 60Hz)
.IPs <ch>\ 
channel number for the sub-channel
.RE
.PD 1
.
.TP
.B surround[=delay]
Decoder for matrix encoded surround sound, works on many 2 channel files.
.PD 0
.RSs
.IPs <delay>
delay time in ms for the rear speakers (0 to 1000) (default: 20)
.RE
.PD 1
.
.TP
.B delay[=ch1:ch2:...]
Delays the sound output.
Specify the delay separately for each channel in milliseconds (floating point
number between 0 and 1000).
.
.TP
.B export[=mmapped_file[:nsamples]]
Exports the incoming signal to other processes using memory mapping (mmap()).
.PD 0
.RSs
.IPs <mmapped_file>
file to map data to (default: ~/.mplayer/\:mplayer-af_export)
.IPs <nsamples>
number of samples per channel (default: 512)
.RE
.PD 1
.
.TP
.B extrastereo[=mul]
Increases the difference between left and right channels to add some
sort of "live" effect to playback.
.PD 0
.RSs
.IPs <mul>
difference coefficient (default: 2.5)
.RE
.PD 1
.
.TP
.B volnorm

from the upper left corner of the bigger image.
.br
.I NOTE:
To get a full list of available video filters, see \-vf help.
.sp 1
Filters are managed in lists.
There are a few commands to manage the filter list.
.
.TP
.B \-vf-add <filter1[,filter2,...]>
Appends the filters given as arguments to the filter list.

14787:346ace66cdb4

To get a full list of available audio filters, see \-af help.
.PP
Available filters are:
.
.TP
.B resample[=srate[:sloppy[:type]]]
Changes the sample rate of the audio stream.
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 necessary.
It only supports the 16-bit little-endian format.
.br
.I NOTE:
With MEncoder, you need to also use \-srate <srate>.
.PD 0
.RSs
.IPs <srate>
output sample frequency in Hz.
The valid range for this parameter is 8000 to 192000.
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.
.IPs <sloppy>
Allow (1) or disallow (0) the output frequency to differ slightly
from the frequency given by <srate> (default: 1).
Can be used if the startup of the playback is extremely slow.
.IPs <type>
Selects which resampling method to use.
.RSss
0: linear interpolation (fast, poor quality especially when upsampling)
.br
1: polyphase filterbank and integer processing
.br
2: polyphase filterbank and floating point processing (slow, best quality)
.REss
.PD 1
.RE
.sp 1
.RS
.I EXAMPLE:
.RE
.PD 0
.RSs
.IPs "mplayer -af resample=44100:0:0"
would set the output frequency of the resample filter to 44100Hz using
exact output frequency scaling and linear interpolation.
.RE
.PD 1
.
.TP
.B lavcresample[=srate[:length[:linear[:count[:cutoff]]]]]
Changes the sample rate of the audio stream to an integer <srate> in Hz.
It only supports the 16-bit little-endian format.
.br
.I NOTE:
With MEncoder, you need to also use \-srate <srate>.
.PD 0
.RSs
.IPs <srate>
the output sample rate

.B hrtf\ \ \ 
Head-related transfer function: Converts multichannel audio to
2 channel output for headphones, preserving the spatiality of the sound.
.
.TP
.B equalizer=[g1:g2:g3:...:g10]
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:
.sp 1
.PD 0
.RS
.IPs "No. frequency"
.IPs "0    31.25 Hz"
.IPs "1    62.50 Hz"
.IPs "2   125.00 Hz"
.IPs "3   250.00 Hz"
.IPs "4   500.00 Hz"
.IPs "5    1.00 kHz"
.IPs "6    2.00 kHz"
.IPs "7    4.00 kHz"
.IPs "8    8.00 kHz"
.IPs "9   16.00 kHz"
.RE
.PD 1
.sp 1
.RS
If the sample rate of the sound being played 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 upsampling the sound
using the resample filter before it reaches this filter.
.RE
.PD 0
.RSs
.IPs <g1>:<g2>:<g3>:...:<g10>
floating point numbers representing the gain in dB
for each frequency band (-12\-12)
.RE
.sp 1
.RS
.I EXAMPLE:
.RE
.RSs
.IPs "mplayer \-af equalizer=11:11:10:5:0:-12:0:5:12:12 media.avi"
Would amplify the sound in the upper and lower frequency region
while canceling it almost completely around 1kHz.
.RE
.PD 1
.
.TP
.B channels=nch[:nr:from1:to1:from2:to2:from3:to3:...]
Can be used for adding, removing, routing and copying audio channels.
If only <nch> 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.
.PD 0
.RSs
.IPs <nch>
number of output channels (1\-6)
.IPs <nr>\ 
number of routes (1\-6)
.IPs <from1:to1:from2:to2:from3:to3:...>
Pairs of numbers between 0 and 5 that define where to route each channel.
.RE
.sp 1
.RS
.I EXAMPLE:
.RE
.RSs
.IPs "mplayer -af channels=4:4:0:1:1:0:2:2:3:3 media.avi"
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.
.IPs "mplayer -af channels=6:4:0:0:0:1:0:2:0:3 media.avi"
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.
.RE
.PD 1
.
.TP
.B format[=format]
Convert between different sample formats.
Automatically enabled when needed by the sound card or another filter.
.PD 0
.RSs
.IPs <format>
Sets the desired format.
The general form is 'sbe', where 's' denotes the sign (either 's' for signed
or 'u' for unsigned), 'b' denotes the number of bits per sample (16, 24 or 32)
and 'e' denotes the endianness ('le' means little-endian, 'be' big-endian
and 'ne' the endianness of the computer MPlayer is running on).
Valid values (amongst others) are: 's16le', 'u32be' and 'u24ne'.
Exceptions to this rule that are also valid format specifiers: u8, s8,
floatle, floatbe, floatne, mulaw, alaw, mpeg2, ac3 and imaadpcm.
.RE
.PD 1
.
.TP
.B volume[=v[:sc]]
Implements 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.
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.
.br
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.
.br
.I NOTE:
This filter is not reentrant and can therefore only be enabled
once for every audio stream.
.PD 0
.RSs
.IPs <v>\ \ 
Sets the desired gain in dB for all channels in the stream
from -200dB to +60dB, where -200dB mutes the sound
completely and +60dB equals a gain of 1000 (default: 0).
.IPs <sc>\ 
Turns soft clipping on (1) or off (0).
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.
.br
.I WARNING:
This feature creates distortion and should be considered a last resort.
.RE
.sp 1
.RS
.I EXAMPLE:
.RE
.RSs
.IPs "mplayer -af volume=10.1:0 media.avi"
would amplify the sound by 10.1dB and hard-clip if the
sound level is too high.
.RE
.PD 1
.
.TP
.B pan=n[:l01:l02:...l10:l11:l12:...ln0:ln1:ln2:...]
Mixes channels arbitrarily.
Basically a combination of the volume and the channels filter
that can be used to down-mix many channels to only a few,
e.g.\& stereo to mono or vary the "width" of the center
speaker in a surround sound system.
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.
An example how to downmix a six-channel file to two channels with
this filter can be found in the examples section near the end.
.PD 0
.RSs
.IPs <n>\ \ 
number of input channels (1\-6)
.IPs <lij>
How much of input channel j is mixed into output channel i (0\-1).
.RE
.sp 1
.RS
.I EXAMPLE:
.RE
.RSs
.IPs "mplayer -af pan=1:0.5:0.5 -channels 1 media.avi"
Would down-mix from stereo to mono.
.IPs "mplayer -af pan=3:1:0:1:0.5:0.5 -channels 3 media.avi"
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 subwoofer for example).
.RE
.PD 1
.
.TP
.B sub[=fc:ch]
Adds a subwoofer channel to the audio stream.
The audio data used for creating the subwoofer 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.
.br
.I Warning:
Disable this filter when you are playing DVDs with Dolby
Digital 5.1 sound, otherwise this filter will disrupt
the sound to the subwoofer.
.PD 0
.RSs
.IPs <fc>\ 
cutoff frequency in Hz for the low-pass filter (20Hz to 300Hz) (default: 60Hz)
For the best result try setting the cutoff frequency as low as possible.
This will improve the stereo or surround sound experience.
.IPs <ch>\ 
Determines the channel number in which to insert the sub-channel audio.
Channel number can be between 0 and 5 (default: 5).
Observe that the number of channels will automatically
be increased to <ch> if necessary.
.RE
.sp 1
.RS
.I EXAMPLE:
.RE
.RSs
.IPs "mplayer -af sub=100:4 -channels 5 media.avi"
would add a sub-woofer channel with a cutoff frequency of
100Hz to output channel 4.
.RE
.PD 1
.
.TP
.B surround[=delay]
Decoder for matrix encoded surround sound like Dolby Surround.
Many files with 2 channel audio actually contain matrixed surround sound.
Requires a sound card supporting at least 4 channels.
.PD 0
.RSs
.IPs <delay>
delay time in ms for the rear speakers (0 to 1000) (default: 20)
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.
.RE
.sp 1
.RS
.I EXAMPLE:
.RE
.RSs
.IPs "mplayer \-af surround=15 \-channels 4 media.avi"
Would add surround sound decoding with 15ms delay for the sound to the
rear speakers.
.RE
.PD 1
.
.TP
.B delay[=ch1:ch2:...]
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.
.PD 0
.RSs
.IPs ch1,ch2,...
The delay in ms that should be imposed on each channel
(floating point number between 0 and 1000).
.RE
.PD 1
.sp 1
.RS
To calculate the required delay for the different channels do as follows:
.IP 1. 3
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
subwoofer (you will not hear the difference anyway).
.IP 2. 3
Subtract the distances s1 to s5 from the maximum distance,
i.e.\& s[i] = max(s) - s[i]; i = 1...5.
.IP 3.
Calculate the required delays in ms as d[i] = 1000*s[i]/342; i = 1...5.
.RE
.PD 0
.sp 1
.RS
.I EXAMPLE:
.RE
.RSs
.IPs "mplayer \-af delay=10.5:10.5:0:0:7:0 media.avi"
Would delay front left and right by 10.5ms, the two rear channels
and the sub by 0ms and the center channel by 7ms.
.RE
.PD 1
.
.TP
.B export[=mmapped_file[:nsamples]]
Exports the incoming signal to other processes using memory mapping (mmap()).
Memory mapped areas contain a header:
.sp 1
.nf
int nch                      /*number of channels*/
int size                     /*buffer size*/
unsigned long long counter   /*Used to keep sync, updated every
                               time new data is exported.*/
.fi
.sp 1
The rest is payload (non-interleaved) 16 bit data.
.PD 0
.RSs
.IPs <mmapped_file>
file to map data to (default: ~/.mplayer/\:mplayer-af_export)
.IPs <nsamples>
number of samples per channel (default: 512)
.RE
.sp 1
.RS
.I EXAMPLE:
.RE
.RSs
.IPs "mplayer \-af export=/tmp/mplayer-af_export:1024 media.avi"
Would export 1024 samples per channel to '/tmp/mplayer-af_export'.
.RE
.PD 1
.
.TP
.B extrastereo[=mul]
(Linearly) increases the difference between left and right channels
which adds some sort of "live" effect to playback.
.PD 0
.RSs
.IPs <mul>
Sets the difference coefficient (default: 2.5).
0.0 means mono sound (average of both channels), with 1.0 sound will be
unchanged, with -1.0 left and right channels will be swapped.
.RE
.PD 1
.
.TP
.B volnorm

from the upper left corner of the bigger image.
.br
.I NOTE:
To get a full list of available video filters, see \-vf help.
.sp 1
Video filters are managed in lists.
There are a few commands to manage the filter list.
.
.TP
.B \-vf-add <filter1[,filter2,...]>
Appends the filters given as arguments to the filter list.