ref: 09be3ef60edd86c77c913b6214ac20f9527de16f
parent: 1790c203bebbe868bd4f5b649b32d5367ea28b4a
author: cbagwell <cbagwell>
date: Fri Aug 31 15:13:12 EDT 2001
Adding some docs for new time parsing.
--- a/sox.1
+++ b/sox.1
@@ -117,6 +117,9 @@
.br
\fBswap\fR [ \fI1 2\fR | \fI1 2 3 4\fR ]
.br
+ \fBsynth\fR [ \fIlength\fR ] \fItype mix\fR [ \fIfreq\fR [ \fI-freq2\fR ]
+ [ \fIoff\fR ] [ \fIph\fR ] [ \fIp1\fR ] [ \fIp2\fR ] [ \fIp3\fR ]
+.br
\fBtrim\fR \fIstart\fR [ \fIlength\fR ]
.br
\fBvibro\fR \fIspeed\fR [ \fIdepth\fR ]
@@ -731,9 +734,13 @@
For fade-outs, the audio data will be truncated at the stop-time and
the volume will be ramped from full volume down to 0 starting at
\fIfade-out-length\fR seconds before the \fIstop-time\fR. No fade-out
-is performed if these options are not specified. All times can be
-specified in seconds, mm:ss.frac, or hh:mm:ss.frac format.
-
+is performed if these options are not specified.
+.br
+All times can be specified in either periods of time or sample counts.
+To specify time periods use the format hh:mm:ss.frac format. To specify
+using sample counts, specify the number of samples and append the letter 's'
+to the sample count (for example 8000s).
+.br
An optional \fItype\fR 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.
.TP 10
filter [ \fIlow\fR ]-[ \fIhigh\fR ] [ \fIwindow-len\fR [ \fIbeta\fR ] ]
@@ -1089,14 +1096,42 @@
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.
+.TP
+synth [ \fIlength\fR ] \fItype mix\fR [ \fIfreq\fR [ \fI-freq2\fR ]
.TP 10
+ [ \fIoff\fR ] [ \fIph\fR ] [ \fIp1\fR ] [ \fIp2\fR ] [ \fIp3\fR ]
+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 determines the length of the output
+audio file.
+.br
+<length> length in sec or hh:mm:ss.frac, 0=inputlength, default=0
+.br
+<type> is sine, square, triangle, sawtooth, trapetz, exp,
+whitenoise, pinknoise, brownnoise, default=sine
+.br
+<mix> is create, mix, amod, default=create
+.br
+<freq> frequency at beginning in Hz, not used for noise..
+.br
+<freq2> frequency at end in Hz, not used for noise..
+<freq/2> can be given as %%n, where 'n' is the number of
+half notes in respect to A (440Hz)
+.br
+<off> Bias (DC-offset) of signal in percent, default=0
+.br
+<ph> phase shift 0..100 shift phase 0..2*Pi, not used for noise..
+.br
+<p1> square: Ton/Toff, triangle+trapetz: rising slope time (0..100)
+.br
+<p2> trapetz: ON time (0..100)
+.br
+<p3> trapetz: falling slope position (0..100)
+.TP 10
trim \fIstart\fR [ \fIlength\fR ]
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 \fIstart\fR location is reached. \fIstart\fR is a floating point number
-that tells the number of seconds to wait before starting. If you know the
-sample number you would like to start at then the seconds can be obtained
-by multiplying (sample # * sample rate).
+the \fIstart\fR location is reached.
.br
The optional \fIlength\fR parameter tells the number of samples to output
after the \fIstart\fR sample and is used to trim off the back side of the
@@ -1103,8 +1138,7 @@
audio data. Using a value of 0 for the \fIstart\fR parameter will allow
trimming off the back side only.
.br
-Both \fIstart\fR and \fIlength\fR can also be specified in mm:ss.frac
-or hh:mm:ss.frac format.
+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.
.TP 10
vibro \fIspeed \fB [ \fIdepth\fB ]
Add the world-famous Fender Vibro-Champ sound
--- a/sox.txt
+++ b/sox.txt
@@ -65,6 +65,8 @@
stat [ -s n ] [ -rms ] [ -v ] [ -d ]
stretch [ factor [ window fade shift fading ]
swap [ 1 2 | 1 2 3 4 ]
+ synth [ length ] type mix [ freq [ -freq2 ]
+ [ off ] [ ph ] [ p1 ] [ p2 ] [ p3 ]
trim start [ length ]
vibro speed [ depth ]
vol gain [ type [ limitergain ] ]
@@ -325,9 +327,9 @@
the list of supported file formats.
.hcom Macintosh HCOM files. These are (apparently)
- Mac FSSD files with some variant of Huffman com�
- pression. The Macintosh has wacky file formats
- and this format handler apparently doesn't
+ Mac FSSD files with some variant of Huffman
+ compression. The Macintosh has wacky file for�
+ mats and this format handler apparently doesn't
handle all the ones it should. Mac users will
need your usual arsenal of file converters to
deal with an HCOM file under Unix or DOS.
@@ -457,11 +459,11 @@
file header, and you will be warned to this
effect. You had better know what you are doing!
Output format options will cause a format con�
- version, and the .wav will written appropri�
- ately. Sox currently can read PCM, ULAW, ALAW,
- MS ADPCM, and IMA (or DVI) ADPCM. It can write
- all of these formats including (NEW!) the ADPCM
- encoding.
+ version, and the .wav will written
+ appropriately. Sox currently can read PCM,
+ ULAW, ALAW, MS ADPCM, and IMA (or DVI) ADPCM.
+ It can write all of these formats including
+ (NEW!) the ADPCM encoding.
.wve Psion 8-bit alaw
These are 8-bit a-law 8khz sound files used on
@@ -670,9 +672,13 @@
from full volume down to 0 starting at fade-out-
length seconds before the stop-time. No fade-
out is performed if these options are not speci�
- fied. All times can be specified in seconds,
- mm:ss.frac, or hh:mm:ss.frac format.
-
+ fied.
+ All times can be specified in either periods of
+ time or sample counts. To specify time periods
+ use the format hh:mm:ss.frac format. To specify
+ using sample counts, specify the number of sam�
+ ples 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
@@ -1037,23 +1043,55 @@
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. Although this effect is used to
+ generate audio data, an input file must be spec�
+ ified. 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, whitenoise, pinknoise, brownnoise,
+ default=sine
+ <mix> is create, mix, amod, default=create
+ <freq> frequency at beginning in Hz, not used
+ for noise..
+ <freq2> frequency at end in Hz, not used for
+ noise.. <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 noise..
+ <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
+ Trim can trim off unwanted audio data from the
beginning and end of the audio file. Audio sam�
ples are not sent to the output stream until the
- start location is reached. start is a floating
- point number that tells the number of seconds to
- wait before starting. If you know the sample
- number you would like to start at then the sec�
- onds can be obtained by multiplying (sample # *
- sample rate).
+ 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 start parame�
ter will allow trimming off the back side only.
- Both start and length can also be specified in
- mm:ss.frac or hh:mm:ss.frac format.
+ 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
@@ -1081,9 +1119,9 @@
changing the phase.
When type is power then a value of 1.0 also
means no change in volume.
- When type is dB the amplitude is changed loga�
- rithmically. 0.0 is constant while +6 doubles
- the amplitude.
+ When type is dB the amplitude is changed
+ logarithmically. 0.0 is constant while +6 dou�
+ bles the amplitude.
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 pre�