ref: 8284ec8c87352c941cbf410978ab5d571c961467
parent: 0fba52507252f37efdb0ecd6534af6103c28d3a5
author: cbagwell <cbagwell>
date: Sat Aug 7 16:20:40 EDT 2004
Remove old effect from docs.
--- a/sox.1
+++ b/sox.1
@@ -122,8 +122,6 @@
.br
\fBspeed\fR [ -c ] \fIfactor\fR
.br
- \fBsplit\fR
-.br
\fBstat\fR [ -s \fIn\fR ] [ -rms ] [ -v ] [ -d ]
.br
\fBstretch\fR [ \fIfactor\fR [ \fIwindow fade shift fading\fR ]
@@ -1132,10 +1130,6 @@
is one octave higher.
0.5 halves speed thus time length doubles and pitch is one octave lower.
If the optional -c parameter is used then the factor is specified in "cents".
-.TP 10
-split
-Turn a mono sample into a stereo sample by copying
-the input channel to the left and right channels.
.TP 10
stat [ \fI-s n\fB ] [\fI-rms\fB ] [ \fI-v\fB ] [ \fI-d\fB ]
Do a statistical check on the input file,
--- a/sox.txt
+++ b/sox.txt
@@ -70,7 +70,6 @@
[ below_periods duration
threshold[ d | % ]]
speed [ -c ] factor
- split
stat [ -s n ] [ -rms ] [ -v ] [ -d ]
stretch [ factor [ window fade shift fading ]
swap [ 1 2 | 1 2 3 4 ]
@@ -605,9 +604,9 @@
dcshift shift [ limitergain ]
DC Shift the audio data, with basic linear amplitude formula.
- This is most useful if your audio data tends to not be cen-
- tered around a value of 0. Shifting it back will allow you
- to get the most volume adjustments without clipping audio
+ This is most useful if your audio data tends to not be
+ centered around a value of 0. Shifting it back will allow
+ you to get the most volume adjustments without clipping audio
data.
The first option is the dcshift value. It is a floating
point number that indicates the amount to shift.
@@ -657,10 +656,10 @@
hh:mm:ss.frac format. To specify using sample counts, spec-
ify the number of samples and append the letter ’s’ to the
sample count (for example 8000s).
- An optional type can be specified to change the type of
- envelope. Choices are q for quarter of a sinewave, h for
- half a sinewave, t for linear slope, l for logarithmic, and p
- for inverted parabola. The default is a linear slope.
+ An optional type can be specified to change the type of enve-
+ lope. Choices are q for quarter of a sinewave, h for half a
+ sinewave, t for linear slope, l for logarithmic, and p for
+ inverted parabola. The default is a linear slope.
filter [ low ]-[ high ] [ window-len [ beta ] ]
Apply a Sinc-windowed lowpass, highpass, or bandpass filter
@@ -942,69 +941,66 @@
octave lower. If the optional -c parameter is used then the
factor is specified in "cents".
- split Turn a mono sample into a stereo sample by copying the input
- channel to the left and right channels.
-
stat [ -s n ] [-rms ] [ -v ] [ -d ]
- Do a statistical check on the input file, and print results
- on the standard error file. Audio data is passed unmodified
- from input to output file unless used along with the -e
+ Do a statistical check on the input file, and print results
+ on the standard error file. Audio data is passed unmodified
+ from input to output file unless used along with the -e
option.
- The "Volume Adjustment:" field in the statistics gives you
- the argument to the -v number which will make the sample as
+ The "Volume Adjustment:" field in the statistics gives you
+ the argument to the -v number which will make the sample as
loud as possible without clipping.
The option -v will print out the "Volume Adjustment:" field’s
- value only and return. This could be of use in scripts to
+ value only and return. This could be of use in scripts to
auto convert the volume.
- The -s n option is used to scale the input data by a given
- factor. The default value of n is the max value of a signed
- long variable (0x7fffffff). Internal effects always work
- with signed long PCM data and so the value should relate to
+ The -s n option is used to scale the input data by a given
+ factor. The default value of n is the max value of a signed
+ long variable (0x7fffffff). Internal effects always work
+ with signed long PCM data and so the value should relate to
this fact.
- The -rms option will convert all output average values to
+ The -rms option will convert all output average values to
root mean square format.
- There is also an optional parameter -d that will print out a
- hex dump of the sound file from the internal buffer that is
- in 32-bit signed PCM data. This is mainly only of use in
- tracking down endian problems that creep in to SoX on cross-
+ There is also an optional parameter -d that will print out a
+ hex dump of the sound file from the internal buffer that is
+ in 32-bit signed PCM data. This is mainly only of use in
+ tracking down endian problems that creep in to SoX on cross-
platform versions.
stretch factor [window fade shift fading]
- Time stretch file by a given factor. Change duration without
- affecting the pitch. factor of stretching: >1.0 lengthen,
- <1.0 shorten duration. window size is in ms. Default is
- 20ms. The fade option, can be "lin". shift ratio, in [0.0
- 1.0]. Default depends on stretch factor. 1.0 to shorten, 0.8
+ Time stretch file by a given factor. Change duration without
+ affecting the pitch. factor of stretching: >1.0 lengthen,
+ <1.0 shorten duration. window size is in ms. Default is
+ 20ms. The fade option, can be "lin". shift ratio, in [0.0
+ 1.0]. Default depends on stretch factor. 1.0 to shorten, 0.8
to lengthen. The fading ratio, in [0.0 0.5]. The amount of a
fade’s default depends on factor and shift.
swap [ 1 2 | 1 2 3 4 ]
- Swap channels in multi-channel sound files. Optionally, you
- may specify the channel order you would like the output in.
- This defaults to output channel 2 and then 1 for stereo and
+ Swap channels in multi-channel sound files. Optionally, you
+ may specify the channel order you would like the output in.
+ This defaults to output channel 2 and then 1 for stereo and
2, 1, 4, 3 for quad-channels. An interesting feature is that
- you may duplicate a given channel by overwriting another.
- This is done by repeating an output channel on the command
- line. For example, swap 2 2 will overwrite channel 1 with
- channel 2’s data; creating a stereo file with both channels
+ you may duplicate a given channel by overwriting another.
+ This is done by repeating an output channel on the command
+ line. For example, swap 2 2 will overwrite channel 1 with
+ channel 2’s data; creating a stereo file with both channels
containing the same audio data.
synth [ length ] type mix [ freq [ -freq2 ]
[ off ] [ ph ] [ p1 ] [ p2 ] [ p3 ]
- The synth effect will generate various types of audio data.
+ The synth effect will generate various types of audio data.
Although this effect is used to generate audio data, an input
- file must be specified. The length of the input audio file
+ file must be specified. The length of the input audio file
determines the length of the output audio file.
<length> length in sec or hh:mm:ss.frac, 0=inputlength,
default=0
- <type> is sine, square, triangle, sawtooth, trapetz, exp,
+ <type> is sine, square, triangle, sawtooth, trapetz, exp,
whitenoise, pinknoise, brownnoise, default=sine
<mix> is create, mix, amod, default=create
<freq> frequency at beginning in Hz, not used for noise..
@@ -1012,66 +1008,66 @@
<freq/2> can be given as %%n, where ’n’ is the number of half
notes in respect to A (440Hz)
<off> Bias (DC-offset) of signal in percent, default=0
- <ph> phase shift 0..100 shift phase 0..2*Pi, not used for
+ <ph> phase shift 0..100 shift phase 0..2*Pi, not used for
noise..
- <p1> square: Ton/Toff, triangle+trapetz: rising slope time
+ <p1> square: Ton/Toff, triangle+trapetz: rising slope time
(0..100)
<p2> trapetz: ON time (0..100)
<p3> trapetz: falling slope position (0..100)
trim start [ length ]
- Trim can trim off unwanted audio data from the beginning and
- end of the audio file. Audio samples are not sent to the
+ Trim can trim off unwanted audio data from the beginning and
+ end of the audio file. Audio samples are not sent to the
output stream until the start location is reached.
- The optional length parameter tells the number of samples to
- output after the start sample and is used to trim off the
- back side of the audio data. Using a value of 0 for the
+ The optional length parameter tells the number of samples to
+ output after the start sample and is used to trim off the
+ back side of the audio data. Using a value of 0 for the
start parameter will allow trimming off the back side only.
- Both options can be specified using either an amount of time
- and an exact count of samples. The format for specifying
- lengths in time is hh:mm:ss.frac. A start value of 1:30.5
- will not start until 1 minute, thirty and 1/2 seconds into
- the audio data. The format for specifying sample counts is
- the number of samples with the letter ’s’ appended to it. A
- value of 8000s will wait until 8000 samples are read before
+ Both options can be specified using either an amount of time
+ and an exact count of samples. The format for specifying
+ lengths in time is hh:mm:ss.frac. A start value of 1:30.5
+ will not start until 1 minute, thirty and 1/2 seconds into
+ the audio data. The format for specifying sample counts is
+ the number of samples with the letter ’s’ appended to it. A
+ value of 8000s will wait until 8000 samples are read before
starting to process audio data.
vibro speed [ depth ]
- Add the world-famous Fender Vibro-Champ sound effect to a
- sound sample by using a sine wave as the volume knob. Speed
- gives the Hertz value of the wave. This must be under 30.
- Depth gives the amount the volume is cut into by the sine
+ Add the world-famous Fender Vibro-Champ sound effect to a
+ sound sample by using a sine wave as the volume knob. Speed
+ gives the Hertz value of the wave. This must be under 30.
+ Depth gives the amount the volume is cut into by the sine
wave, ranging 0.0 to 1.0 and defaulting to 0.5.
vol gain [ type [ limitergain ] ]
- The vol effect is much like the command line option -v. It
- allows you to adjust the volume of an input file and allows
- you to specify the adjustment in relation to amplitude,
- power, or dB. If type is not specified then it defaults to
+ The vol effect is much like the command line option -v. It
+ allows you to adjust the volume of an input file and allows
+ you to specify the adjustment in relation to amplitude,
+ power, or dB. If type is not specified then it defaults to
amplitude.
- When type is amplitude then a linear change of the amplitude
- is performed based on the gain. Therefore, a value of 1.0
- will keep the volume the same, 0.0 to < 1.0 will cause the
- volume to decrease and values of > 1.0 will cause the volume
- to increase. Beware of clipping audio data when the gain is
+ When type is amplitude then a linear change of the amplitude
+ is performed based on the gain. Therefore, a value of 1.0
+ will keep the volume the same, 0.0 to < 1.0 will cause the
+ volume to decrease and values of > 1.0 will cause the volume
+ to increase. Beware of clipping audio data when the gain is
greater then 1.0. A negative value performs the same adjust-
ment while also changing the phase.
- When type is power then a value of 1.0 also means no change
+ When type is power then a value of 1.0 also means no change
in volume.
- When type is dB the amplitude is changed logarithmically.
+ When type is dB the amplitude is changed logarithmically.
0.0 is constant while +6 doubles the amplitude.
- An optional limitergain value can be specified and should be
+ An optional limitergain value can be specified and should be
a value much less then 1.0 (ie 0.05 or 0.02) and is used only
- on peaks to prevent clipping. Not specifying this parameter
- will cause no limiter to be used. In verbose mode, this
- effect will display the percentage of audio data that needed
+ on peaks to prevent clipping. Not specifying this parameter
+ will cause no limiter to be used. In verbose mode, this
+ effect will display the percentage of audio data that needed
to be limited.
BUGS
- The syntax is horrific. Thats the breaks when trying to handle all
+ The syntax is horrific. Thats the breaks when trying to handle all
things from the command line.
- Please report any bugs found in this version of SoX to Chris Bagwell
+ Please report any bugs found in this version of SoX to Chris Bagwell
(cbagwell@users.sourceforge.net)
FILES
@@ -1079,9 +1075,9 @@
play(1), rec(1), soxexam(1)
NOTICES
- The version of SoX that accompanies this manual page is support by
+ The version of SoX that accompanies this manual page is support by
Chris Bagwell (cbagwell@users.sourceforge.net). Please refer any ques-
- tions regarding it to this address. You may obtain the latest version
+ tions regarding it to this address. You may obtain the latest version
at the the web site http://sox.sourceforge.net/
AUTHOR