[MPlayer-DOCS] CVS: main/DOCS/man/en mplayer.1,1.893,1.894
Diego Biurrun CVS
syncmail at mplayerhq.hu
Thu Feb 24 12:00:48 CET 2005
CVS change done by Diego Biurrun CVS
Update of /cvsroot/mplayer/main/DOCS/man/en
In directory mail:/var2/tmp/cvs-serv15177/DOCS/man/en
Modified Files:
mplayer.1
Log Message:
Move audio filter descriptions to the man page.
Index: mplayer.1
===================================================================
RCS file: /cvsroot/mplayer/main/DOCS/man/en/mplayer.1,v
retrieving revision 1.893
retrieving revision 1.894
diff -u -r1.893 -r1.894
--- mplayer.1 24 Feb 2005 01:55:18 -0000 1.893
+++ mplayer.1 24 Feb 2005 11:00:43 -0000 1.894
@@ -3507,15 +3507,57 @@
Available filters are:
.
.TP
-.B resample[=srate[:sloppy][:type]]
-Changes the sample rate of the audio stream to an integer srate in Hz.
+.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.
+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
@@ -3544,17 +3586,93 @@
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.
+.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]
-Change the current sample format.
+Convert between different sample formats.
+Automatically enabled when needed by the sound card or another filter.
.PD 0
.RSs
.IPs <format>
@@ -3564,72 +3682,207 @@
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.
+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]
-Select the output volume level.
-This filter is not reentrant and can therefore only be enabled once for every
-audio stream.
+.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).
+from -200dB to +60dB, where -200dB mutes the sound
+completely and +60dB equals a gain of 1000 (default: 0).
.IPs <sc>\
-Enable soft clipping.
+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, 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.
+.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.
+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]
-Add subwoofer channel.
+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 for low-pass filter (20Hz to 300Hz) (default: 60Hz)
+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>\
-channel number for the sub-channel
+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, works on many 2 channel files.
+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 output.
-Specify the delay separately for each channel in milliseconds (floating point
-number between 0 and 1000).
+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>
@@ -3637,16 +3890,26 @@
.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]
-Increases the difference between left and right channels to add some
-sort of "live" effect to playback.
+(Linearly) increases the difference between left and right channels
+which adds some sort of "live" effect to playback.
.PD 0
.RSs
.IPs <mul>
-difference coefficient (default: 2.5)
+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
.
@@ -3703,7 +3966,7 @@
.I NOTE:
To get a full list of available video filters, see \-vf help.
.sp 1
-Filters are managed in lists.
+Video filters are managed in lists.
There are a few commands to manage the filter list.
.
.TP
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