# HG changeset patch # User diego # Date 1042294159 0 # Node ID 5b73c925436ed06106970a25b1eb4de6f296b63e # Parent c9cbfb9d720c44145b3ff069d3cb582a878bcad7 Further libaf documentation by Anders with some more updates by me. diff -r c9cbfb9d720c -r 5b73c925436e DOCS/documentation.html --- a/DOCS/documentation.html Sat Jan 11 14:02:23 2003 +0000 +++ b/DOCS/documentation.html Sat Jan 11 14:09:19 2003 +0000 @@ -197,6 +197,8 @@
would set the output frequency of the resample filter to 11025Hz and downmix the audio to 1 channel using the pan filter.
-Most filters respond to the -v
switch, which makes the filters
- print out status messages.
The overall execution of the filter layer is controlled using the
-af-adv
switch. This switch has two suboptions:
force
0
1
2
3
4
5
6
7
list
The filter layer is also affected by the following generic switches: + +
-v
-channels
-srate
-format
srate
srate <8-192>
sloppy
srate
. This switch can be
- used if the startup of the playback is extremely slow.fast
type <0-2>
0
and 2
that
+ selects which resampling method to use. Here 0
represents
+ linear interpolation as resampling method, 1
represents
+ resampling using a poly-phase filter-bank and integer processing and
+ 2
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 2
but this method also suffers from
+ the highest CPU load.Example:
@@ -268,19 +306,19 @@
itself if not needed. The number of switches is dynamic:
nch
nch <1-6>
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.nr
nr <1-6>
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.from1:to1:from2:to2:from3:to3...
0
and 5
that define
+ where each channel should be routed.If only nch
is given the default routing is used, it works as
@@ -311,11 +349,12 @@
needed by the sound card or another filter.
bps
bps <number>
1
, 2
or 4
and denotes the
+ number of bytes per sample. This switch is required, leaving it empty
+ results in a runtime error.f
f <format>
alaw
, mulaw
or
imaadpcm
, float
or int
,
@@ -377,9 +416,9 @@
background is gone. This filter has two switches:
v
v <-200 - +60>
-200
and +60
+ which represents the volume level in dB. The default level is -10dB.c
This filter is a 10 octave band graphic equalizer, implemented using 10 IIR +
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:
@@ -427,12 +466,12 @@ that band. This problem can be worked around by up-sampling the sound using the resample filter before it reaches this filter. -This filter has 10 parameters:
+This filter has 10 parameters:
g1:g2:g3...g10
-12 and +12
+ representing the gain in dB for each frequency band.
Example:
@@ -457,14 +496,15 @@
the number of output channels:
nch
nch <1-6>
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.l00:l01:l02:..l10:l11:l12:...ln0:ln1:ln2:...
l[i][j]
determines
- how much of input channel j is mixed into output channel i.0
and 1
.
+ l[i][j]
determines how much of input channel j is mixed into
+ output channel i.Example 1:
@@ -480,6 +520,65 @@
example).
This 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 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:
+ +fc <20-300>
ch <0-5>
0
and 5
which
+ determines the channel number in which to insert the sub-channel audio.
+ The default is channel number 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.
+ +This filter is a decoder for matrix encoded surround sound. 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:
+ +d <0-1000>
0
and
+ 1000
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.Example:
+ mplayer -af surround=15 -channels 4 media.avi
would add a surround sound decoding with 15ms delay for the sound to the rear + speakers.
+ + + +