ref: 430ffc96aada94d90a4f95d5b5b52cac9f2dc849
parent: c4ef72a989e0e73ea494b25fe862b306c1eab81f
author: cbagwell <cbagwell>
date: Mon Sep 3 21:58:11 EDT 2001
Updating more docs with new effects and handlers.
--- a/Changelog
+++ b/Changelog
@@ -48,7 +48,7 @@
to create a test sound CD for testing audio equipment.
o Ben Last added a new program that uses libst and will merge two
seperate audio files into a single file with multiple channels.
- Written for sox 12.16 so doesn't work with 12.17.2 yet.
+ This was merged into the standard sox.c file by cbagwell.
o Andreas Menke fixed some problems with the speed effect and
how effects were drained. Also improved the usage of printf()'s
to use stderr.
--- a/Makefile.dos
+++ b/Makefile.dos
@@ -17,7 +17,7 @@
wav.obj wve.obj
EOBJ = avg.obj band.obj bandpass.obj breject.obj btrworth.obj chorus.obj \
- compand.obj copy.obj cut.obj dcshift.obj deemphas.obj earwax.o \
+ compand.obj copy.obj dcshift.obj deemphas.obj earwax.o \
echo.obj echos.obj fade.obj filter.obj flanger.obj highp.obj \
highpass.obj lowp.obj lowpass.obj map.obj mask.obj phaser.obj \
pick.obj pitch.obj pan.obj polyphase.obj rate.obj resample.obj \
--- a/Makefile.gcc
+++ b/Makefile.gcc
@@ -34,7 +34,7 @@
tx16w.o voc.o wav.o wve.o
EOBJ = avg.o band.o bandpass.o breject.o btrworth.o chorus.o compand.o \
- copy.o cut.o dcshift.o deemphas.o earwax.o echo.o echos.o fade.o \
+ copy.o dcshift.o deemphas.o earwax.o echo.o echos.o fade.o \
filter.o flanger.o highp.o highpass.o lowp.o lowpass.o map.o \
mask.o pan.o phaser.o pick.o pitch.o polyphas.o rate.o \
resample.o reverb.o reverse.o silence.o speed.o split.o \
--- a/README
+++ b/README
@@ -45,7 +45,7 @@
o Band-pass filters
o Band-reject filter
o Chorus effect
- o Cut out loop samples
+ o DCShift audio. Useful to get the best volume adjustments.
o Move sound stage of CD audio to in front of you (for headphone use)
o Add an echo
o Add a sequence of echos
@@ -64,6 +64,7 @@
changes using real signal theory!
o Apply a reverb effect
o Reverse the sound samples (to search for Satanic messages ;-)
+ o Trim off silence from the beginning of end of a file.
o Change the speed of samples being played (like speeding up the motor
on a tape recorder)
o Convert from mono to stereo
@@ -80,7 +81,7 @@
This is release 12.17, Patchlevel 2 of the Sound Tools.
SoX was originally written and maintained by Lance Norskog but
unfortunetly he has stopped maintaining it since 1995. I, Chris
-Bagwell (cbagwell@sprynet.com), have started maintaining it since
+Bagwell (cbagwell@users.sourceforge.net), have started maintaining it since
1996 to the present.
Caveats:
--- a/TODO
+++ b/TODO
@@ -1,5 +1,5 @@
People are encouraged to pick some of these and implement it. Send
-all patches to cbagwell@sprynet.com.
+all patches to cbagwell@users.sourceforge.net.
o Add in manual page entries for silence and dcshift.
--- a/cut.c
+++ /dev/null
@@ -1,121 +1,0 @@
-
-/*
- * July 5, 1991
- * Copyright 1991 Lance Norskog And Sundry Contributors
- * This source code is freely redistributable and may be used for
- * any purpose. This copyright notice must be maintained.
- * Lance Norskog And Sundry Contributors are not responsible for
- * the consequences of using this software.
- */
-
-/*
- * Sound Tools cut effect file.
- *
- * Pull loop #n from a looped sound file.
- * Not finished, don't use it yet.
- */
-
-#include <math.h>
-#include "st.h"
-
-/* Private data for SKEL file */
-typedef struct cutstuff {- int which; /* Loop # to pull */
- int where; /* current sample # */
- ULONG start; /* first wanted sample */
- ULONG end; /* last wanted sample + 1 */
-} *cut_t;
-
-/*
- * Process options
- */
-int st_cut_getopts(effp, n, argv)
-eff_t effp;
-int n;
-char **argv;
-{- cut_t cut = (cut_t) effp->priv;
-
- /* parse it */
- cut->which = 0; /* for now */
- return (ST_SUCCESS);
-}
-
-/*
- * Prepare processing.
- */
-int st_cut_start(effp)
-eff_t effp;
-{- cut_t cut = (cut_t) effp->priv;
- /* nothing to do */
-
- cut->where = 0;
- cut->start = effp->loops[0].start;
- cut->end = effp->loops[0].start + effp->loops[0].length;
- return (ST_SUCCESS);
-}
-
-/*
- * Processed signed long samples from ibuf to obuf.
- * Return number of samples processed.
- */
-
-int st_cut_flow(effp, ibuf, obuf, isamp, osamp)
-eff_t effp;
-LONG *ibuf, *obuf;
-LONG *isamp, *osamp;
-{- cut_t cut = (cut_t) effp->priv;
- int len, done;
-
- len = ((*isamp > *osamp) ? *osamp : *isamp);
- if ((cut->where + len <= cut->start) ||
- (cut->where >= cut->end)) {- *isamp = len;
- *osamp = 0;
- cut->where += len;
- return (ST_SUCCESS);
- }
- *isamp = len; /* We will have processed all inputs */
- if (cut->where < cut->start) {- /* skip */
- ibuf += cut->start - cut->where;
- len -= cut->start - cut->where;
- }
- if (cut->where + len >= cut->end) {- /* shorten */
- len = cut->end - cut->where;
- }
- for(done = 0; done < len; done++) {- *obuf++ = *ibuf++;
- }
- *osamp = len;
- return (ST_SUCCESS);
-}
-
-/*
- * Drain out remaining samples if the effect generates any.
- */
-
-int st_cut_drain(effp, obuf, osamp)
-eff_t effp;
-LONG *obuf;
-LONG *osamp;
-{- *osamp = 0;
- return (ST_SUCCESS);
-}
-
-/*
- * Do anything required when you stop reading samples.
- * (free allocated memory, etc.)
- */
-int st_cut_stop(effp)
-eff_t effp;
-{- /* nothing to do */
- return (ST_SUCCESS);
-}
-
-
--- a/sox.1
+++ b/sox.1
@@ -58,7 +58,7 @@
.br
\fBcopy\fR
.br
- \fBcut\fR
+ \fBdcshift\fR \fIshift\fR [ \fIlimitergain\fR ]
.br
\fBdeemph\fR
.br
@@ -404,6 +404,13 @@
Allows 8bit linear, 16bit linear, A-Law, u-law
in mono and stereo.
.TP 10
+.B .nul
+Null file handler. This is a fake file hander that act as if its reading
+a stream of 0's from a while or fake writing output to a file. This
+is not a very useful file handler in most cases. It might be useful in
+some scripts were you do not want to read or write from a real file
+but would like to specify a filename for consistency.
+.TP 10
.B .ogg
Ogg Vorbis Compressed Audio.
.br
@@ -696,8 +703,16 @@
This is the default effect if both files have the same
sampling rate.
.TP 10
-cut \fIloopnumber
-Extract loop #N from a sample.
+dcshift \fIshift\fR [ \fIlimitergain\fR ]
+DC Shift the audio data, with basic linear amplitudate formula.
+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.
+.br
+The first option is the \fIdcshift\fR value. It is a floating point number that
+indicates the amount to shift.
+.br
+An option limtergain value can be specified as well. It should have a value much less then 1.0 and is used only on peaks to prevent clipping.
.TP 10
deemph
Apply a treble attenuation shelving filter to samples in
@@ -1211,6 +1226,6 @@
.BR soxexam(1)
.SH NOTICES
The version of Sox that accompanies this manual page is support by
-Chris Bagwell (cbagwell@sprynet.com). Please refer any questions
+Chris Bagwell (cbagwell@users.sourceforge.net). Please refer any questions
regarding it to this address. You may obtain the latest version at the
-the web site http://home.sprynet.com/~cbagwell/sox.html
+the web site http://sox.sourceforge.net/
--- a/sox.txt
+++ b/sox.txt
@@ -34,7 +34,7 @@
in-dB1,out-dB1[,in-dB2,out-dB2...]
[ gain [ initial-volume [ delay ] ] ]
copy
- cut
+ dcshift shift [ limitergain ]
deemph
earwax
echo gain-in gain-out delay decay [ delay decay ... ]
@@ -344,65 +344,75 @@
8bit linear, 16bit linear, A-Law, u-law in mono
and stereo.
+ .nul Null file handler. This is a fake file hander
+ that act as if its reading a stream of 0's from
+ a while or fake writing output to a file. This
+ is not a very useful file handler in most cases.
+ It might be useful in some scripts were you do
+ not want to read or write from a real file but
+ would like to specify a filename for consis�
+ tency.
+
.ogg Ogg Vorbis Compressed Audio.
Ogg Vorbis is a open, patent-free codec designed
- for compressing music and streaming audio. It
- is similar to MP3, VQF, AAC, and other lossy
+ for compressing music and streaming audio. It
+ is similar to MP3, VQF, AAC, and other lossy
formats. sox can decode all types of Ogg Vorbis
- files, but can only encode at 128 kbps. Decod�
- ing is somewhat CPU intensive and encoding is
+ files, but can only encode at 128 kbps. Decod�
+ ing is somewhat CPU intensive and encoding is
very CPU intensive.
- Ogg Vorbis in sox is optional and requires
+ Ogg Vorbis in sox is optional and requires
access to external Ogg Vorbis libraries. To see
- if there is support for Ogg Vorbis run sox -h
+ if there is support for Ogg Vorbis run sox -h
and look for it under the list of supported file
formats as "vorbis".
ossdsp OSS /dev/dsp device driver
This is a pseudo-file type and can be optionally
- compiled into Sox. Run sox -h to see if you
- have support for this file type. When this
- driver is used it allows you to open up the OSS
- /dev/dsp file and configure it to use the same
- data format as passed in to /fBSoX. It works
- for both playing and recording sound samples.
- When playing sound files it attempts to set up
- the OSS driver to use the same format as the
- input file. It is suggested to always override
- the output values to use the highest quality
+ compiled into Sox. Run sox -h to see if you
+ have support for this file type. When this
+ driver is used it allows you to open up the OSS
+ /dev/dsp file and configure it to use the same
+ data format as passed in to /fBSoX. It works
+ for both playing and recording sound samples.
+ When playing sound files it attempts to set up
+ the OSS driver to use the same format as the
+ input file. It is suggested to always override
+ the output values to use the highest quality
samples your sound card can handle. Example: -t
ossdsp -w -s /dev/dsp
.sf IRCAM Sound Files.
- Sound Files are used by academic music software
- such as the CSound package, and the MixView
+ Sound Files are used by academic music software
+ such as the CSound package, and the MixView
sound sample editor.
.sph
- SPHERE (SPeech HEader Resources) is a file for�
+ SPHERE (SPeech HEader Resources) is a file for�
mat defined by NIST (National Institute of Stan�
- dards and Technology) and is used with speech
- audio. SoX can read these files when they con�
- tain ulaw and PCM data. It will ignore any
- header information that says the data is com�
- pressed using shorten compression and will treat
- the data as either ulaw or PCM. This will allow
- SoX and the command line shorten program to be
- ran together using pipes to uncompress the data
- and then pass the result to SoX for processing.
+ dards and Technology) and is used with speech
+ audio. SoX can read these files when they con�
+ tain ulaw and PCM data. It will ignore any
+ header information that says the data is
+ compressed using shorten compression and will
+ treat the data as either ulaw or PCM. This will
+ allow SoX and the command line shorten program
+ to be ran together using pipes to uncompress the
+ data and then pass the result to SoX for pro�
+ cessing.
.smp Turtle Beach SampleVision files.
- SMP files are for use with the PC-DOS package
- SampleVision by Turtle Beach Softworks. This
- package is for communication to several MIDI
- samplers. All sample rates are supported by the
- package, although not all are supported by the
- samplers themselves. Currently loop points are
+ SMP files are for use with the PC-DOS package
+ SampleVision by Turtle Beach Softworks. This
+ package is for communication to several MIDI
+ samplers. All sample rates are supported by the
+ package, although not all are supported by the
+ samplers themselves. Currently loop points are
ignored.
.snd
- Under DOS this file format is the same as the
- .sndt format. Under all other platforms it is
+ Under DOS this file format is the same as the
+ .sndt format. Under all other platforms it is
the same as the .au format.
.sndt SoundTool files.
@@ -410,132 +420,132 @@
sunau Sun /dev/audio device driver
This is a pseudo-file type and can be optionally
- compiled into Sox. Run sox -h to see if you
- have support for this file type. When this
- driver is used it allows you to open up a Sun
+ compiled into Sox. Run sox -h to see if you
+ have support for this file type. When this
+ driver is used it allows you to open up a Sun
/dev/audio file and configure it to use the same
- data type as passed in to Sox. It works for
- both playing and recording sound samples. When
- playing sound files it attempts to set up the
+ data type as passed in to Sox. It works for
+ both playing and recording sound samples. When
+ playing sound files it attempts to set up the
audio driver to use the same format as the input
- file. It is suggested to always override the
+ file. It is suggested to always override the
output values to use the highest quality samples
- your hardware can handle. Example: -t sunau -w
+ your hardware can handle. Example: -t sunau -w
-s /dev/audio or -t sunau -U -c 1 /dev/audio for
older sun equipment.
.txw Yamaha TX-16W sampler.
- A file format from a Yamaha sampling keyboard
- which wrote IBM-PC format 3.5" floppies. Han�
+ A file format from a Yamaha sampling keyboard
+ which wrote IBM-PC format 3.5" floppies. Han�
dles reading of files which do not have the sam�
- ple rate field set to one of the expected by
- looking at some other bytes in the attack/loop
- length fields, and defaulting to 33kHz if the
+ ple rate field set to one of the expected by
+ looking at some other bytes in the attack/loop
+ length fields, and defaulting to 33kHz if the
sample rate is still unknown.
.vms More info to come.
- Used to compress speech audio for applications
+ Used to compress speech audio for applications
such as voice mail.
.voc Sound Blaster VOC files.
- VOC files are multi-part and contain silence
- parts, looping, and different sample rates for
- different chunks. On input, the silence parts
- are filled out, loops are rejected, and sample
- data with a new sample rate is rejected.
- Silence with a different sample rate is gener�
- ated appropriately. On output, silence is not
+ VOC files are multi-part and contain silence
+ parts, looping, and different sample rates for
+ different chunks. On input, the silence parts
+ are filled out, loops are rejected, and sample
+ data with a new sample rate is rejected.
+ Silence with a different sample rate is gener�
+ ated appropriately. On output, silence is not
detected, nor are impossible sample rates.
vorbis See .ogg format.
.wav Microsoft .WAV RIFF files.
- These appear to be very similar to IFF files,
- but not the same. They are the native sound
+ These appear to be very similar to IFF files,
+ but not the same. They are the native sound
file format of Windows. (Obviously, Windows was
- of such incredible importance to the computer
- industry that it just had to have its own sound
+ of such incredible importance to the computer
+ industry that it just had to have its own sound
file format.) Normally .wav files have all for�
- matting information in their headers, and so do
- not need any format options specified for an
- input file. If any are, they will override the
- file header, and you will be warned to this
+ matting information in their headers, and so do
+ not need any format options specified for an
+ input file. If any are, they will override the
+ 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
+ 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.
.wve Psion 8-bit alaw
- These are 8-bit a-law 8khz sound files used on
+ These are 8-bit a-law 8khz sound files used on
the Psion palmtop portable computer.
.raw Raw files (no header).
- The sample rate, size (byte, word, etc), and
+ The sample rate, size (byte, word, etc), and
encoding (signed, unsigned, etc.) of the sample
- file must be given. The number of channels
+ file must be given. The number of channels
defaults to 1.
.ub, .sb, .uw, .sw, .ul, .al, .sl
- These are several suffices which serve as a
- shorthand for raw files with a given size and
- encoding. Thus, ub, sb, uw, sw, ul and sl cor�
- respond to "unsigned byte", "signed byte",
- "unsigned word", "signed word", "ulaw" (byte),
- "alaw" (byte), and "signed long". The sample
- rate defaults to 8000 hz if not explicitly set,
- and the number of channels (as always) defaults
- to 1. There are lots of Sparc samples floating
- around in u-law format with no header and fixed
- at a sample rate of 8000 hz. (Certain sound
+ These are several suffices which serve as a
+ shorthand for raw files with a given size and
+ encoding. Thus, ub, sb, uw, sw, ul and sl cor�
+ respond to "unsigned byte", "signed byte",
+ "unsigned word", "signed word", "ulaw" (byte),
+ "alaw" (byte), and "signed long". The sample
+ rate defaults to 8000 hz if not explicitly set,
+ and the number of channels (as always) defaults
+ to 1. There are lots of Sparc samples floating
+ around in u-law format with no header and fixed
+ at a sample rate of 8000 hz. (Certain sound
management software cheerfully ignores the head�
- ers.) Similarly, most Mac sound files are in
+ ers.) Similarly, most Mac sound files are in
unsigned byte format with a sample rate of 11025
or 22050 hz.
- .auto This is a ``meta-type'': specifying this type
- for an input file triggers some code that tries
- to guess the real type by looking for magic
- words in the header. If the type can't be
- guessed, the program exits with an error mes�
- sage. The input must be a plain file, not a
+ .auto This is a ``meta-type'': specifying this type
+ for an input file triggers some code that tries
+ to guess the real type by looking for magic
+ words in the header. If the type can't be
+ guessed, the program exits with an error mes�
+ sage. The input must be a plain file, not a
pipe. This type can't be used for output files.
EFFECTS
Multiple effects may be applied to the audio data by spec�
- ifying them one after another at the end of the command
+ ifying them one after another at the end of the command
line.
avg [ -l | -r | -f | -b | n,n,...,n ]
- Reduce the number of channels by averaging the
- samples, or duplicate channels to increase the
- number of channels. This effect is automati�
- cally used when the number of input channels
+ Reduce the number of channels by averaging the
+ samples, or duplicate channels to increase the
+ number of channels. This effect is automati�
+ cally used when the number of input channels
differ from the number of output channels. When
- reducing the number of channels it is possible
- to manually specify the avg effect and use the
- -l, -r, -f, or -b options to select only the
- left, right, front, or back channel(s) for the
- output instead of averaging the channels. The
- -f and -b options maintain left/right stereo
+ reducing the number of channels it is possible
+ to manually specify the avg effect and use the
+ -l, -r, -f, or -b options to select only the
+ left, right, front, or back channel(s) for the
+ output instead of averaging the channels. The
+ -f and -b options maintain left/right stereo
separation; use the avg effect twice to select a
single channel.
The avg effect can also be invoked with up to 16
double-precision numbers, which specify the pro�
- portion of each input channel that is to be
- mixed into each output channel. In two-channel
+ portion of each input channel that is to be
+ mixed into each output channel. In two-channel
mode, 4 numbers are given: l->l, l->r, r->l, and
- r->r, respectively. In four-channel mode, the
- first 4 numbers give the proportions for the
- left-front output channel, as follows: lf->lf,
+ r->r, respectively. In four-channel mode, the
+ first 4 numbers give the proportions for the
+ left-front output channel, as follows: lf->lf,
rf->lf, lb->lf, and rb->rf. The next 4 give the
right-front output in the same order, then left-
back and right-back.
- It is also possible to use the 16 numbers to
+ It is also possible to use the 16 numbers to
expand or reduce the channel count; just specify
0 for unused channels. Finally, if fewer than 4
numbers are given, certain special abbreviations
@@ -542,24 +552,24 @@
may be invoked; see the source code for details.
band [ -n ] center [ width ]
- Apply a band-pass filter. The frequency
+ Apply a band-pass filter. The frequency
response drops logarithmically around the center
- frequency. The width gives the slope of the
- drop. The frequencies at center + width and
- center - width will be half of their original
+ frequency. The width gives the slope of the
+ drop. The frequencies at center + width and
+ center - width will be half of their original
amplitudes. Band defaults to a mode oriented to
pitched signals, i.e. voice, singing, or instru�
- mental music. The -n (for noise) option uses
- the alternate mode for un-pitched signals.
- Warning: -n introduces a power-gain of about
- 11dB in the filter, so beware of output clip�
+ mental music. The -n (for noise) option uses
+ the alternate mode for un-pitched signals.
+ Warning: -n introduces a power-gain of about
+ 11dB in the filter, so beware of output clip�
ping. Band introduces noise in the shape of the
filter, i.e. peaking at the center frequency and
- settling around it. See filter for a bandpass
+ settling around it. See filter for a bandpass
effect with steeper shoulders.
bandpass frequency bandwidth
- Butterworth bandpass filter. Description coming
+ Butterworth bandpass filter. Description coming
soon!
bandreject frequency bandwidth
@@ -569,10 +579,10 @@
chorus gain-in gain-out delay decay speed depth
-s | -t [ delay decay speed depth -s | -t ... ]
- Add a chorus to a sound sample. Each quadtuple
- delay/decay/speed/depth gives the delay in mil�
- liseconds and the decay (relative to gain-in)
- with a modulation speed in Hz using depth in
+ Add a chorus to a sound sample. Each quadtuple
+ delay/decay/speed/depth gives the delay in mil�
+ liseconds and the decay (relative to gain-in)
+ with a modulation speed in Hz using depth in
milliseconds. The modulation is either sinodial
(-s) or triangular (-t). Gain-out is the volume
of the output.
@@ -582,55 +592,66 @@
in-dB1,out-dB1[,in-dB2,out-dB2...]
[gain [initial-volume [delay ] ] ]
- Compand (compress or expand) the dynamic range
- of a sample. The attack and decay time specify
- the integration time over which the absolute
- value of the input signal is integrated to
+ Compand (compress or expand) the dynamic range
+ of a sample. The attack and decay time specify
+ the integration time over which the absolute
+ value of the input signal is integrated to
determine its volume; attacks refer to increases
- in volume and decays refer to decreases. Where
- more than one pair of attack/decay parameters
- are specified, each channel is treated sepa�
- rately and the number of pairs must agree with
- the number of input channels. The second
- parameter is a list of points on the compander's
- transfer function specified in dB relative to
+ in volume and decays refer to decreases. Where
+ more than one pair of attack/decay parameters
+ are specified, each channel is treated sepa�
+ rately and the number of pairs must agree with
+ the number of input channels. The second param�
+ eter is a list of points on the compander's
+ transfer function specified in dB relative to
the maximum possible signal amplitude. The
- input values must be in a strictly increasing
+ input values must be in a strictly increasing
order but the transfer function does not have to
be monotonically rising. The special value -inf
- may be used to indicate that the input volume
- should be associated output volume. The points
+ may be used to indicate that the input volume
+ should be associated output volume. The points
-inf,-inf and 0,0 are assumed; the latter may be
overridden, but the former may not.
The third (optional) parameter is a postprocess�
- ing gain in dB which is applied after the com�
- pression has taken place; the fourth (optional)
+ ing gain in dB which is applied after the com�
+ pression has taken place; the fourth (optional)
parameter is an initial volume to be assumed for
- each channel when the effect starts. This per�
- mits the user to supply a nominal level ini�
- tially, so that, for example, a very large gain
- is not applied to initial signal levels before
- the companding action has begun to operate: it
- is quite probable that in such an event, the
- output would be severely clipped while the com�
+ each channel when the effect starts. This per�
+ mits the user to supply a nominal level ini�
+ tially, so that, for example, a very large gain
+ is not applied to initial signal levels before
+ the companding action has begun to operate: it
+ is quite probable that in such an event, the
+ output would be severely clipped while the com�
pander gain properly adjusts itself.
- The fifth (optional) parameter is a delay in
- seconds. The input signal is analyzed immedi�
+ The fifth (optional) parameter is a delay in
+ seconds. The input signal is analyzed immedi�
ately to control the compander, but it is
delayed before being fed to the volume adjuster.
- Specifying a delay approximately equal to the
- attack/decay times allows the compander to
- effectively operate in a "predictive" rather
+ Specifying a delay approximately equal to the
+ attack/decay times allows the compander to
+ effectively operate in a "predictive" rather
than a reactive mode.
copy Copy the input file to the output file. This is
- the default effect if both files have the same
+ the default effect if both files have the same
sampling rate.
- cut loopnumber
- Extract loop #N from a sample.
+ dcshift shift [ limitergain ]
+ DC Shift the audio data, with basic linear
+ amplitudate formula. 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 clip�
+ ping audio data.
+ The first option is the dcshift value. It is a
+ floating point number that indicates the amount
+ to shift.
+ An option limtergain value can be specified as
+ well. It should have a value much less then 1.0
+ and is used only on peaks to prevent clipping.
deemph Apply a treble attenuation shelving filter to
samples in audio cd format. The frequency
@@ -834,12 +855,12 @@
rate Translate input sampling rate to output sampling
rate via linear interpolation to the Least Com�
mon Multiple of the two sampling rates. This is
- the default effect if the two files have differ�
- ent sampling rates and the preview options was
- specified. This is fast but noisy: the spectrum
- of the original sound will be shifted upwards
- and duplicated faintly when up-translating by a
- multiple.
+ the default effect if the two files have
+ different sampling rates and the preview options
+ was specified. This is fast but noisy: the
+ spectrum of the original sound will be shifted
+ upwards and duplicated faintly when up-translat�
+ ing by a multiple.
Lerp-ing is acceptable for cheap 8-bit sound
hardware, but for CD-quality sound you should
@@ -1177,10 +1198,10 @@
NOTICES
The version of Sox that accompanies this manual page is
- support by Chris Bagwell (cbagwell@sprynet.com). Please
- refer any questions regarding it to this address. You may
- obtain the latest version at the the web site
- http://home.sprynet.com/~cbagwell/sox.html
+ support by Chris Bagwell (cbagwell@users.sourceforge.net).
+ Please refer any questions regarding it to this address.
+ You may obtain the latest version at the the web site
+ http://sox.sourceforge.net/
--- a/src/Makefile.in
+++ b/src/Makefile.in
@@ -48,7 +48,7 @@
wav.o wve.o
EOBJ = avg.o band.o bandpass.o breject.o btrworth.o chorus.o compand.o \
- copy.o cut.o dcshift.o deemphas.o earwax.o echo.o echos.o \
+ copy.o dcshift.o deemphas.o earwax.o echo.o echos.o \
fade.o filter.o flanger.o highp.o highpass.o lowp.o lowpass.o \
map.o mask.o pan.o phaser.o pick.o pitch.o polyphas.o \
rate.o resample.o reverb.o reverse.o silence.o speed.o split.o \
--- a/src/dcshift.c
+++ b/src/dcshift.c
@@ -30,12 +30,9 @@
#define ONE ((DCSHIFT_FLOAT)(1.0e0))
#define TWENTY ((DCSHIFT_FLOAT)(20.0e0))
-#define DCSHIFT_USAGE "Usage: dcshift shift [ limitergain ]"
-// The following is dieing on a solaris gcc 2.8.1 compiler!?!!
-#if 0
-#define DCSHIFT_USAGE2 " The peak limiter has a gain much less than 1.0 (ie 0.05 or 0.02) which is only"
-#define DCSHIFT_USAGE3 " used on peaks to prevent clipping. (default is no limiter)"
-#endif
+#define DCSHIFT_USAGE "Usage: dcshift shift [ limitergain ]\n" \
+"The peak limiter has a gain much less than 1.0 (ie 0.05 or 0.02) which is only\n" \
+"used on peaks to prevent clipping. (default is no limiter)"
typedef struct {
DCSHIFT_FLOAT dcshift; /* DC shift. */
@@ -60,7 +57,13 @@
dcs_t dcs = (dcs_t) effp->priv;
dcs->dcshift = ONE; /* default is no change */
dcs->uselimiter = 0; /* default is no limiter */
-
+
+ if (n < 1)
+ {
+ st_fail(DCSHIFT_USAGE);
+ return ST_EOF;
+ }
+
if (n && (!sscanf(argv[0], DCSHIFT_FLOAT_SCAN, &dcs->dcshift)))
{
st_fail(DCSHIFT_USAGE);
--- a/src/handlers.c
+++ b/src/handlers.c
@@ -529,11 +529,6 @@
extern int st_copy_flow();
extern int st_copy_stop();
-extern int st_cut_getopts();
-extern int st_cut_start();
-extern int st_cut_flow();
-extern int st_cut_stop();
-
extern int st_dcshift_getopts();
extern int st_dcshift_start();
extern int st_dcshift_flow();
@@ -741,9 +736,6 @@
st_compand_drain, st_compand_stop},
{"copy", ST_EFF_MCHAN, st_copy_getopts, st_copy_start, st_copy_flow,
- st_null_drain, st_nothing},
- {"cut", ST_EFF_MCHAN, - st_cut_getopts, st_cut_start, st_cut_flow,
st_null_drain, st_nothing},
{"dcshift", ST_EFF_MCHAN, st_dcshift_getopts, st_dcshift_start, st_dcshift_flow,
--- a/src/play.in
+++ b/src/play.in
@@ -65,7 +65,7 @@
# loop over arguments
while [ $# -ne 0 ]; do
case "$1" in
- avg|band|bandpass|bandreject|chorus|compand|copy|cut|deemph|earwax|echo|echos|fade|filter|flanger|highp|highpass|lowp|lowpass|map|mask|pan|phaser|pick|pitch|polyphase|rate|resample|reverb|reverse|speed|split|stat|stretch|swap|trim|vibro|vol)
+ avg|band|bandpass|bandreject|chorus|compand|copy|cut|deemph|earwax|echo|echos|fade|filter|flanger|highp|highpass|lowp|lowpass|map|mask|pan|phaser|pick|pitch|polyphase|rate|resample|reverb|reverse|silence|speed|split|stat|stretch|swap|trim|vibro|vol)
effects="$@"
break
;;
--- a/src/sox.c
+++ b/src/sox.c
@@ -523,6 +523,7 @@
}
}
+#ifdef SOXMIX
static int compare_input(ft_t ft1, ft_t ft2)
{if (ft1->info.rate != ft2->info.rate)
@@ -536,6 +537,7 @@
return ST_SUCCESS;
}
+#endif
/*
* Process input file -> effect table -> output file