mirror of https://github.com/mpv-player/mpv
543 lines
22 KiB
ReStructuredText
543 lines
22 KiB
ReStructuredText
.. _audio_filters:
|
|
|
|
AUDIO FILTERS
|
|
=============
|
|
|
|
Audio filters allow you to modify the audio stream and its properties. The
|
|
syntax is:
|
|
|
|
--af=<filter1[=parameter1:parameter2:...],filter2,...>
|
|
Setup a chain of audio filters.
|
|
|
|
*NOTE*: To get a full list of available audio filters, see ``--af=help``.
|
|
|
|
Audio filters are managed in lists. There are a few commands to manage the
|
|
filter list.
|
|
|
|
--af-add=<filter1[,filter2,...]>
|
|
Appends the filters given as arguments to the filter list.
|
|
|
|
--af-pre=<filter1[,filter2,...]>
|
|
Prepends the filters given as arguments to the filter list.
|
|
|
|
--af-del=<index1[,index2,...]>
|
|
Deletes the filters at the given indexes. Index numbers start at 0,
|
|
negative numbers address the end of the list (-1 is the last).
|
|
|
|
--af-clr
|
|
Completely empties the filter list.
|
|
|
|
Available filters are:
|
|
|
|
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 16-bit integer and float in native-endian
|
|
format as input.
|
|
|
|
<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.
|
|
<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.
|
|
<type>
|
|
Select which resampling method to use.
|
|
|
|
:0: linear interpolation (fast, poor quality especially when
|
|
upsampling)
|
|
:1: polyphase filterbank and integer processing
|
|
:2: polyphase filterbank and floating point processing
|
|
(slow, best quality)
|
|
|
|
*EXAMPLE*:
|
|
|
|
``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.
|
|
|
|
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 native-endian format.
|
|
|
|
<srate>
|
|
the output sample rate
|
|
<length>
|
|
length of the filter with respect to the lower sampling rate (default:
|
|
16)
|
|
<linear>
|
|
if 1 then filters will be linearly interpolated between polyphase
|
|
entries
|
|
<count>
|
|
log2 of the number of polyphase entries (..., 10->1024, 11->2048,
|
|
12->4096, ...) (default: 10->1024)
|
|
<cutoff>
|
|
cutoff frequency (0.0-1.0), default set depending upon filter length
|
|
|
|
lavcac3enc[=tospdif[:bitrate[:minchn]]]
|
|
Encode multi-channel audio to AC-3 at runtime using libavcodec. Supports
|
|
16-bit native-endian input format, maximum 6 channels. The output is
|
|
big-endian when outputting a raw AC-3 stream, native-endian when
|
|
outputting to S/PDIF. The output sample rate of this filter is same with
|
|
the input sample rate. When input sample rate is 48kHz, 44.1kHz, or 32kHz,
|
|
this filter directly use it. Otherwise a resampling filter is
|
|
auto-inserted before this filter to make the input and output sample rate
|
|
be 48kHz. You need to specify ``--channels=N`` to make the decoder decode
|
|
audio into N-channel, then the filter can encode the N-channel input to
|
|
AC-3.
|
|
|
|
<tospdif>
|
|
Output raw AC-3 stream if zero or not set, output to S/PDIF for
|
|
passthrough when <tospdif> is set non-zero.
|
|
<bitrate>
|
|
The bitrate to encode the AC-3 stream. Set it to either 384 or 384000
|
|
to get 384kbits.
|
|
|
|
Valid values: 32, 40, 48, 56, 64, 80, 96, 112, 128,
|
|
160, 192, 224, 256, 320, 384, 448, 512, 576, 640.
|
|
|
|
Default bitrate is based on the input channel number:
|
|
|
|
:1ch: 96
|
|
:2ch: 192
|
|
:3ch: 224
|
|
:4ch: 384
|
|
:5ch: 448
|
|
:6ch: 448
|
|
|
|
<minchn>
|
|
If the input channel number is less than <minchn>, the filter will
|
|
detach itself (default: 5).
|
|
|
|
sweep[=speed]
|
|
Produces a sine sweep.
|
|
|
|
<0.0-1.0>
|
|
Sine function delta, use very low values to hear the sweep.
|
|
|
|
sinesuppress[=freq:decay]
|
|
Remove a sine at the specified frequency. Useful to get rid of the 50/60Hz
|
|
noise on low quality audio equipment. It probably only works on mono input.
|
|
|
|
<freq>
|
|
The frequency of the sine which should be removed (in Hz) (default:
|
|
50)
|
|
<decay>
|
|
Controls the adaptivity (a larger value will make the filter adapt to
|
|
amplitude and phase changes quicker, a smaller value will make the
|
|
adaptation slower) (default: 0.0001). Reasonable values are around
|
|
0.001.
|
|
|
|
bs2b[=option1:option2:...]
|
|
Bauer stereophonic to binaural transformation using ``libbs2b``. Improves
|
|
the headphone listening experience by making the sound similar to that
|
|
from loudspeakers, allowing each ear to hear both channels and taking into
|
|
account the distance difference and the head shadowing effect. It is
|
|
applicable only to 2 channel audio.
|
|
|
|
fcut=<300-1000>
|
|
Set cut frequency in Hz.
|
|
feed=<10-150>
|
|
Set feed level for low frequencies in 0.1*dB.
|
|
profile=<value>
|
|
Several profiles are available for convenience:
|
|
|
|
:default: will be used if nothing else was specified (fcut=700,
|
|
feed=45)
|
|
:cmoy: Chu Moy circuit implementation (fcut=700, feed=60)
|
|
:jmeier: Jan Meier circuit implementation (fcut=650, feed=95)
|
|
|
|
If fcut or feed options are specified together with a profile, they will
|
|
be applied on top of the selected profile.
|
|
|
|
hrtf[=flag]
|
|
Head-related transfer function: Converts multichannel audio to 2 channel
|
|
output for headphones, preserving the spatiality of the sound.
|
|
|
|
==== ===================================
|
|
Flag Meaning
|
|
==== ===================================
|
|
m matrix decoding of the rear channel
|
|
s 2-channel matrix decoding
|
|
0 no matrix decoding (default)
|
|
==== ===================================
|
|
|
|
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:
|
|
|
|
=== ==========
|
|
No. frequency
|
|
=== ==========
|
|
0 31.25 Hz
|
|
1 62.50 Hz
|
|
2 125.00 Hz
|
|
3 250.00 Hz
|
|
4 500.00 Hz
|
|
5 1.00 kHz
|
|
6 2.00 kHz
|
|
7 4.00 kHz
|
|
8 8.00 kHz
|
|
9 16.00 kHz
|
|
=== ==========
|
|
|
|
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.
|
|
|
|
<g1>:<g2>:<g3>:...:<g10>
|
|
floating point numbers representing the gain in dB for each frequency
|
|
band (-12-12)
|
|
|
|
*EXAMPLE*:
|
|
|
|
``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.
|
|
|
|
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.
|
|
|
|
<nch>
|
|
number of output channels (1-8)
|
|
<nr>
|
|
number of routes (1-8)
|
|
<from1:to1:from2:to2:from3:to3:...>
|
|
Pairs of numbers between 0 and 7 that define where to route each
|
|
channel.
|
|
|
|
*EXAMPLE*:
|
|
|
|
``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.
|
|
|
|
``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.
|
|
|
|
format[=format]
|
|
Convert between different sample formats. Automatically enabled when
|
|
needed by the sound card or another filter. See also ``--format``.
|
|
|
|
<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.
|
|
|
|
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.
|
|
|
|
This filter has a second feature: It measures the overall maximum sound
|
|
level and prints out that level when MPlayer exits. This feature currently
|
|
only works with floating-point data, use e.g. ``--af-adv=force=5``, or use
|
|
``--af=stats``.
|
|
|
|
*NOTE*: This filter is not reentrant and can therefore only be enabled
|
|
once for every audio stream.
|
|
|
|
<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).
|
|
<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.
|
|
|
|
*WARNING*: This feature creates distortion and should be considered a
|
|
last resort.
|
|
|
|
*EXAMPLE*:
|
|
|
|
``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.
|
|
|
|
pan=n[:L00: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.
|
|
|
|
<n>
|
|
number of output channels (1-8)
|
|
<Lij>
|
|
How much of input channel i is mixed into output channel j (0-1). So
|
|
in principle you first have n numbers saying what to do with the first
|
|
input channel, then n numbers that act on the second input channel
|
|
etc. If you do not specify any numbers for some input channels, 0 is
|
|
assumed.
|
|
|
|
*EXAMPLE*:
|
|
|
|
``mplayer --af=pan=1:0.5:0.5 media.avi``
|
|
Would down-mix from stereo to mono.
|
|
|
|
``mplayer --af=pan=3:1:0:0.5:0:1:0.5 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).
|
|
|
|
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.
|
|
|
|
*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.
|
|
|
|
<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.
|
|
<ch>
|
|
Determines the channel number in which to insert the sub-channel
|
|
audio. Channel number can be between 0 and 7 (default: 5). Observe
|
|
that the number of channels will automatically be increased to <ch> if
|
|
necessary.
|
|
|
|
*EXAMPLE*:
|
|
|
|
``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.
|
|
|
|
center
|
|
Creates a center channel from the front channels. May currently be low
|
|
quality as it does not implement a high-pass filter for proper extraction
|
|
yet, but averages and halves the channels instead.
|
|
|
|
<ch>
|
|
Determines the channel number in which to insert the center channel.
|
|
Channel number can be between 0 and 7 (default: 5). Observe that the
|
|
number of channels will automatically be increased to <ch> if
|
|
necessary.
|
|
|
|
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.
|
|
|
|
<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 should be
|
|
set to 15ms if d1 <= d2 and to 15 + 5*(d1-d2) if d1 > d2.
|
|
|
|
*EXAMPLE*:
|
|
|
|
``mplayer --af=surround=15 --channels=4 media.avi``
|
|
Would add surround sound decoding with 15ms delay for the sound to the
|
|
rear speakers.
|
|
|
|
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.
|
|
|
|
ch1,ch2,...
|
|
The delay in ms that should be imposed on each channel (floating point
|
|
number between 0 and 1000).
|
|
|
|
To calculate the required delay for the different channels do as follows:
|
|
|
|
1. 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).
|
|
|
|
2. Subtract the distances s1 to s5 from the maximum distance, i.e.
|
|
``s[i] = max(s) - s[i]; i = 1...5``.
|
|
|
|
3. Calculate the required delays in ms as ``d[i] = 1000*s[i]/342; i =
|
|
1...5``.
|
|
|
|
*EXAMPLE*:
|
|
|
|
``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.
|
|
|
|
export[=mmapped_file[:nsamples]]
|
|
Exports the incoming signal to other processes using memory mapping
|
|
(``mmap()``). Memory mapped areas contain a header:
|
|
|
|
| int nch /\* number of channels \*/
|
|
| int size /\* buffer size \*/
|
|
| unsigned long long counter /\* Used to keep sync, updated every time new data is exported. \*/
|
|
|
|
The rest is payload (non-interleaved) 16 bit data.
|
|
|
|
<mmapped_file>
|
|
file to map data to (default: ``~/.mplayer/mplayer-af_export``)
|
|
<nsamples>
|
|
number of samples per channel (default: 512)
|
|
|
|
*EXAMPLE*:
|
|
|
|
``mplayer --af=export=/tmp/mplayer-af_export:1024 media.avi``
|
|
Would export 1024 samples per channel to ``/tmp/mplayer-af_export``.
|
|
|
|
extrastereo[=mul]
|
|
(Linearly) increases the difference between left and right channels which
|
|
adds some sort of "live" effect to playback.
|
|
|
|
<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.
|
|
|
|
volnorm[=method:target]
|
|
Maximizes the volume without distorting the sound.
|
|
|
|
<method>
|
|
Sets the used method.
|
|
|
|
1
|
|
Use a single sample to smooth the variations via the standard
|
|
weighted mean over past samples (default).
|
|
2
|
|
Use several samples to smooth the variations via the standard
|
|
weighted mean over past samples.
|
|
|
|
<target>
|
|
Sets the target amplitude as a fraction of the maximum for the sample
|
|
type (default: 0.25).
|
|
|
|
ladspa=file:label[:controls...]
|
|
Load a LADSPA (Linux Audio Developer's Simple Plugin API) plugin. This
|
|
filter is reentrant, so multiple LADSPA plugins can be used at once.
|
|
|
|
<file>
|
|
Specifies the LADSPA plugin library file. If ``LADSPA_PATH`` is set,
|
|
it searches for the specified file. If it is not set, you must supply
|
|
a fully specified pathname.
|
|
<label>
|
|
Specifies the filter within the library. Some libraries contain only
|
|
one filter, but others contain many of them. Entering 'help' here,
|
|
will list all available filters within the specified library, which
|
|
eliminates the use of 'listplugins' from the LADSPA SDK.
|
|
<controls>
|
|
Controls are zero or more floating point values that determine the
|
|
behavior of the loaded plugin (for example delay, threshold or gain).
|
|
In verbose mode (add ``-v`` to the MPlayer command line), all
|
|
available controls and their valid ranges are printed. This eliminates
|
|
the use of 'analyseplugin' from the LADSPA SDK.
|
|
|
|
karaoke
|
|
Simple voice removal filter exploiting the fact that voice is usually
|
|
recorded with mono gear and later 'center' mixed onto the final audio
|
|
stream. Beware that this filter will turn your signal into mono. Works
|
|
well for 2 channel tracks; do not bother trying it on anything but 2
|
|
channel stereo.
|
|
|
|
scaletempo[=option1:option2:...]
|
|
Scales audio tempo without altering pitch, optionally synced to playback
|
|
speed (default).
|
|
|
|
This works by playing 'stride' ms of audio at normal speed then consuming
|
|
'stride*scale' ms of input audio. It pieces the strides together by
|
|
blending 'overlap'% of stride with audio following the previous stride. It
|
|
optionally performs a short statistical analysis on the next 'search' ms
|
|
of audio to determine the best overlap position.
|
|
|
|
scale=<amount>
|
|
Nominal amount to scale tempo. Scales this amount in addition to
|
|
speed. (default: 1.0)
|
|
stride=<amount>
|
|
Length in milliseconds to output each stride. Too high of value will
|
|
cause noticable skips at high scale amounts and an echo at low scale
|
|
amounts. Very low values will alter pitch. Increasing improves
|
|
performance. (default: 60)
|
|
overlap=<percent>
|
|
Percentage of stride to overlap. Decreasing improves performance.
|
|
(default: .20)
|
|
search=<amount>
|
|
Length in milliseconds to search for best overlap position. Decreasing
|
|
improves performance greatly. On slow systems, you will probably want
|
|
to set this very low. (default: 14)
|
|
speed=<tempo|pitch|both|none>
|
|
Set response to speed change.
|
|
|
|
tempo
|
|
Scale tempo in sync with speed (default).
|
|
pitch
|
|
Reverses effect of filter. Scales pitch without altering tempo.
|
|
Add ``[ speed_mult 0.9438743126816935`` and ``] speed_mult
|
|
1.059463094352953`` to your ``input.conf`` to step by musical
|
|
semi-tones.
|
|
|
|
*WARNING*: Loses sync with video.
|
|
both
|
|
Scale both tempo and pitch.
|
|
none
|
|
Ignore speed changes.
|
|
|
|
*EXAMPLE*:
|
|
|
|
``mplayer --af=scaletempo --speed=1.2 media.ogg``
|
|
Would playback media at 1.2x normal speed, with audio at normal pitch.
|
|
Changing playback speed, would change audio tempo to match.
|
|
|
|
``mplayer --af=scaletempo=scale=1.2:speed=none --speed=1.2 media.ogg``
|
|
Would playback media at 1.2x normal speed, with audio at normal pitch,
|
|
but changing playback speed has no effect on audio tempo.
|
|
|
|
``mplayer --af=scaletempo=stride=30:overlap=.50:search=10 media.ogg``
|
|
Would tweak the quality and performace parameters.
|
|
|
|
``mplayer --af=format=floatne,scaletempo media.ogg``
|
|
Would make scaletempo use float code. Maybe faster on some platforms.
|
|
|
|
``mplayer --af=scaletempo=scale=1.2:speed=pitch audio.ogg``
|
|
Would playback audio file at 1.2x normal speed, with audio at normal
|
|
pitch. Changing playback speed, would change pitch, leaving audio
|
|
tempo at 1.2x.
|
|
|
|
stats
|
|
Collects and prints statistics about the audio stream, especially the
|
|
volume. These statistics are especially intended to help adjusting the
|
|
volume while avoiding clipping. The volumes are printed in dB and
|
|
compatible with the volume audio filter.
|