ref: bf36071fc0b18250336fa12789b87ec49e160b87
parent: b65a5923a78a1e96ae1b97843950fc3eabe7c6a6
author: cbagwell <cbagwell>
date: Thu Sep 7 21:01:54 EDT 2000
Final update of docs for 12.17
--- a/libst.txt
+++ b/libst.txt
@@ -12,7 +12,7 @@
cc file.c -o file libst.a
DESCRIPTION
- Sound Tools is a library of sound sample file format read-
+ Sound Tools is a library of sound sample file format read�
ers/writers and sound effects processors.
Sound Tools includes skeleton C files to assist you in
@@ -32,17 +32,17 @@
The Sound Tools formats and effects operate on an internal
buffer format of signed 32-bit longs. The data processing
routines are called with buffers of these samples, and
- buffer sizes which refer to the number of samples pro-
+ buffer sizes which refer to the number of samples pro�
cessed, not the number of bytes. File readers translate
the input samples to signed longs and return the number of
- longs read. For example, data in linear signed byte for-
+ longs read. For example, data in linear signed byte for�
mat is left-shifted 24 bits.
This does cause problems in processing the data. For
example:
*obuf++ = (*ibuf++ + *ibuf++)/2;
- would not mix down left and right channels into one mono-
- phonic channel, because the resulting samples would over-
+ would not mix down left and right channels into one mono�
+ phonic channel, because the resulting samples would over�
flow 32 bits. Instead, the ``avg'' effects must use:
*obuf++ = *ibuf++/2 + *ibuf++/2;
@@ -52,10 +52,10 @@
rear.
FORMATS
- A format is responsible for translating between sound sam-
+ A format is responsible for translating between sound sam�
ple files and an internal buffer. The internal buffer is
store in signed longs with a fixed sampling rate. The
- format operates from two data structures: a format struc-
+ format operates from two data structures: a format struc�
ture, and a private structure.
@@ -76,7 +76,7 @@
number of sound channels. It also contains other state
information: whether the sample file needs to be byte-
swapped, whether fseek() will work, its suffix, its file
- stream pointer, its format pointer, and the private struc-
+ stream pointer, its format pointer, and the private struc�
ture for the format .
The private area is just a preallocated data array for the
@@ -98,7 +98,7 @@
read Given a buffer and a length: read up
to that many samples, transform them
into signed long integers, and copy
- them into the buffer. Return the num-
+ them into the buffer. Return the num�
ber of samples actually read.
stopread Do what needs to be done.
@@ -108,7 +108,7 @@
be done.
write Given a buffer and a length: copy that
- many samples out of the buffer, con-
+ many samples out of the buffer, con�
vert them from signed longs to the
appropriate data, and write them to
the file. If it can't write out all
@@ -144,7 +144,7 @@
and output data sizes. It processes
the input buffer into the output
buffer, and sets the size variables to
- the numbers of samples actually pro-
+ the numbers of samples actually pro�
cessed. It is under no obligation to
fill the output buffer.
--- a/sox.txt
+++ b/sox.txt
@@ -76,11 +76,12 @@
stat [ -s n ] [ -rms ] [ -v ] [ -d ]
stretch [ factor [ window fade shift fading ]
swap [ 1 2 | 1 2 3 4 ]
+ trim start [ length ]
vibro speed [ depth ]
- vol gain [ type ]
+ vol gain [ type [ limitergain ] ]
DESCRIPTION
- SoX is a command line program that can convert most popu-
+ SoX is a command line program that can convert most popu�
lar audio files to most other popular audio file formats.
It can optionally change the audio sample data type and
apply one or more sound effects to the file during this
@@ -88,7 +89,7 @@
There are two types of audio files formats that SoX can
work with. The first are self-describing file formats.
- These contain a header that completely describe the char-
+ These contain a header that completely describe the char�
acteristics of the audio data that follows.
The second type are headerless data, or sometimes called
@@ -96,13 +97,13 @@
the command line so that it knows what type of data it
contains.
- Audio data can usually be totally described by four char-
+ Audio data can usually be totally described by four char�
acteristics:
rate The sample rate is in samples per second. For
example, CD sample rates are at 44100.
- data size The precision the data is stored in. Most popu-
+ data size The precision the data is stored in. Most popu�
lar are 8-bit bytes or 16-bit words.
data encoding
@@ -126,7 +127,6 @@
-
July 24, 2000 2
@@ -145,19 +145,19 @@
Format options:
- Format options effect the audio samples that they immedi-
+ Format options effect the audio samples that they immedi�
ately preceed. If they are placed before the input file
name then they effect the input data. If they are placed
- before the output file name then they will effect the out-
+ before the output file name then they will effect the out�
put data. By taking advantage of this, you can override a
input file's corrupted header or produce an output file
that is totally different style then the input file. It
is also how sox is informed about the format of raw input
- data.:w
+ data.
-t filetype
gives the type of the sound sample file. Useful
- when file extension is not standard or for spec-
+ when file extension is not standard or for spec�
ifying the .auto file type.
-r rate Gives the sample rate in Hertz of the file. To
@@ -172,7 +172,7 @@
-s/-u/-U/-A/-a/-i/-g
The sample data encoding is signed linear (2's
- complement), unsigned linear, U-law (logarith-
+ complement), unsigned linear, U-law (logarith�
mic), A-law (logarithmic), ADPCM, IMA_ADPCM, or
GSM.
U-law (actually shorthand for mu-law) and A-law
@@ -187,7 +187,7 @@
fidelity is not as important. When uncompressed
it has roughly the precision of 16-bit PCM
audio. Popular version of ADPCM include G.726,
- MS ADPCM, and IMA ADPCM. The -a flag has dif-
+ MS ADPCM, and IMA ADPCM. The -a flag has dif�
ferent meanings in different file handlers. In
.wav files it represents MS ADPCM files, in all
@@ -207,7 +207,7 @@
simpler and slightly lower fidelity than
Microsoft's flavor of ADPCM. IMA ADPCM is also
called DVI ADPCM.
- GSM is a standard used for telephone sound com-
+ GSM is a standard used for telephone sound com�
pression in European countries and its gaining
popularity because of its quality. It usually
is CPU intensive to work with GSM audio data.
@@ -215,7 +215,7 @@
-b/-w/-l/-f/-d/-D
The sample data size is in bytes, 16-bit words,
32-bit longwords, 32-bit floats, 64-bit double
- floats, or 80-bit IEEE floats. Floats and dou-
+ floats, or 80-bit IEEE floats. Floats and dou�
ble floats are in native machine format.
-x The sample data is in XINU format; that is, it
@@ -241,7 +241,7 @@
-e When used after the input filename (so that it
applies to the output file) it allows you to
avoid giving an output filename and will not
- produce an output file. It will apply any spec-
+ produce an output file. It will apply any spec�
ified effects to the input file. This is mainly
useful with the stat effect but can be used with
others.
@@ -255,7 +255,7 @@
a different number of channels and a different
rate than the input file. Currently, this
defaults to using the rate effect instead of the
- resample for sample rate changes.
+ resample effect for sample rate changes.
@@ -279,7 +279,7 @@
this option without causing audio data be be
clipped.
- -V Print a description of processing phases. Use-
+ -V Print a description of processing phases. Use�
ful for figuring out exactly how sox is mangling
your sound samples.
@@ -299,13 +299,13 @@
The following file formats are supported:
- .8svx Amiga 8SVX musical instrument description for-
+ .8svx Amiga 8SVX musical instrument description for�
mat.
.aiff AIFF files used on Apple IIc/IIgs and SGI.
Note: the AIFF format supports only one SSND
chunk. It does not support multiple sound
- chunks, or the 8SVX musical instrument descrip-
+ chunks, or the 8SVX musical instrument descrip�
tion format. AIFF files are multimedia archives
and can have multiple audio and picture chunks.
You may need a separate archiver to work with
@@ -334,16 +334,16 @@
SoX(1) SoX(1)
- The AVR format is produced by a number of com-
+ The AVR format is produced by a number of com�
mercial packages on the Mac.
.cdr CD-R
- CD-R files are used in mastering music on Com-
+ CD-R files are used in mastering music on Com�
pact Disks. The audio data on a CD-R disk is a
raw audio file with a format of stereo 16-bit
signed samples at a 44khz sample rate. There is
a special blocking/padding oddity at the end of
- the audio file and is why it needs its own han-
+ the audio file and is why it needs its own han�
dler.
.cvs Continuously Variable Slope Delta modulation
@@ -353,13 +353,13 @@
.dat Text Data files
These files contain a textual representation of
the sample data. There is one line at the
- beginning that contains the sample rate. Subse-
+ beginning that contains the sample rate. Subse�
quent lines contain two numeric data items: the
time since the beginning of the first sample and
the sample value. Values are normalized so that
the maximum and minimum are 1.00 and -1.00.
This file format can be used to create data
- files for external programs such as FFT analyz-
+ files for external programs such as FFT analyz�
ers or graph routines. SoX can also convert a
file in this format back into one of the other
file formats.
@@ -366,8 +366,8 @@
.gsm GSM 06.10 Lossy Speech Compression
A standard for compressing speech which is used
- in the Global Standard for Mobil telecommunica-
- tions (GSM). Its good for its purpose, shrink-
+ in the Global Standard for Mobil telecommunica�
+ tions (GSM). Its good for its purpose, shrink�
ing audio data size, but it will introduce lots
of noise when a given sound sample is encoded
and decoded multiple times. This format is used
@@ -374,14 +374,14 @@
by some voice mail applications. It is rather
CPU intensive.
GSM in sox is optional and requires access to an
- external GSM library. To see if there is sup-
+ external GSM library. To see if there is sup�
port for gsm run sox -h and look for it under
the list of supported file formats.
.hcom Macintosh HCOM files. These are (apparently)
- Mac FSSD files with some variant of Huffman com-
+ Mac FSSD files with some variant of Huffman com�
pression. The Macintosh has wacky file formats
- and this format handler apparently doesn't han-
+ and this format handler apparently doesn't han�
dle 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.
@@ -426,12 +426,12 @@
sound sample editor.
.sph
- SPHERE (SPeech HEader Resources) is a file for-
- mat defined by NIST (National Institute of Stan-
+ 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-
+ 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-
+ 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
@@ -484,8 +484,8 @@
.txw Yamaha TX-16W sampler.
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-
+ 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
@@ -501,7 +501,7 @@
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-
+ Silence with a different sample rate is gener�
ated appropriately. On output, silence is not
detected, nor are impossible sample rates.
@@ -511,14 +511,14 @@
file format of Windows. (Obviously, Windows was
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-
+ 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
effect. You had better know what you are doing!
- Output format options will cause a format con-
- version, and the .wav will written appropri-
+ Output format options will cause a format con�
+ version, and the .wav will written appropri�
ately. Sox currently can read PCM, ULAW, ALAW,
@@ -549,7 +549,7 @@
.ub, .sb, .uw, .sw, .ul, .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-
+ encoding. Thus, ub, sb, uw, sw, ul and sl cor�
respond to "unsigned byte", "signed byte",
"unsigned word", "signed word", "ulaw" (byte),
and "signed long". The sample rate defaults to
@@ -566,12 +566,12 @@
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-
+ 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-
+ Multiple effects may be applied to the audio data by spec�
ifying them one after another at the end of the command
line.
@@ -578,13 +578,13 @@
avg [ -l | -r ]
Reduce the number of channels by averaging the
samples, or duplicate channels to increase the
- number of channels. This effect is automati-
+ 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 and -r options to select only the left or
- right channel for the output instead of averag-
+ right channel for the output instead of averag�
ing the two channels.
@@ -605,11 +605,11 @@
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-
+ 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-
+ 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
@@ -620,7 +620,7 @@
soon!
bandreject frequency bandwidth
- Butterworth bandreject filter. Description com-
+ Butterworth bandreject filter. Description com�
ing soon!
chorus gain-in gain-out delay decay speed depth
@@ -627,7 +627,7 @@
-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-
+ 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
@@ -646,7 +646,7 @@
determine its volume. Where more than one pair
of attack/decay parameters are specified, each
channel is treated separately and the number of
- pairs must agree with the number of input chan-
+ pairs must agree with the number of input chan�
nels. The second parameter is a list of points
on the compander's transfer function specified
in dB relative to the maximum possible signal
@@ -664,9 +664,9 @@
SoX(1) SoX(1)
- function does not have to be monotonically ris-
+ function does not have to be monotonically ris�
ing. The special value -inf may be used to
- indicate that the input volume should be associ-
+ indicate that the input volume should be associ�
ated output volume. The points -inf,-inf and
0,0 are assumed; the latter may be overridden,
but the former may not. The third (optional)
@@ -678,9 +678,9 @@
to supply a nominal level initially, 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 proba-
+ action has begun to operate: it is quite proba�
ble that in such an event, the output would be
- severely clipped while the compander gain prop-
+ severely clipped while the compander gain prop�
erly adjusts itself.
copy Copy the input file to the output file. This is
@@ -692,7 +692,7 @@
deemph Apply a treble attenuation shelving filter to
samples in audio cd format. The frequency
- response of pre-emphasized recordings is recti-
+ response of pre-emphasized recordings is recti�
fied. The filtering is defined in the standard
document ISO 908.
@@ -738,7 +738,7 @@
windows a more gradual cutoff.
The beta, if unspecified, defaults to 16. This
- selects a Kaiser window. You can select a Nut-
+ selects a Kaiser window. You can select a Nut�
tall window by specifying anything <= 2.0 here.
For more discussion of beta, look under the
resample effect.
@@ -746,7 +746,7 @@
flanger gain-in gain-out delay decay speed < -s | -t >
Add a flanger to a sound sample. Each triple
- delay/decay/speed gives the delay in millisec-
+ delay/decay/speed gives the delay in millisec�
onds and the decay (relative to gain-in) with a
modulation speed in Hz. The modulation is
either sinodial (-s) or triangular (-t). Gain-
@@ -760,7 +760,7 @@
for a highpass effect with sharper cutoff.
highpass frequency
- Butterworth highpass filter. Description com-
+ Butterworth highpass filter. Description com�
ming soon!
lowp frequency
@@ -774,7 +774,7 @@
Butterworth lowpass filter. Description coming
soon!
- map Display a list of loops in a sample, and miscel-
+ map Display a list of loops in a sample, and miscel�
laneous loop info.
mask Add "masking noise" to signal. This effect
@@ -804,8 +804,8 @@
number of input channels is different then the
number of output channels then this effect tries
to intelligently handle this. For instance, if
- the input contains 1 channel and the output con-
- tains 2 channels, then it will create the miss-
+ the input contains 1 channel and the output con�
+ tains 2 channels, then it will create the miss�
ing channel itself. The direction is a value
from -1.0 to 1.0. -1.0 represents far left and
1.0 represents far right. Numbers in between
@@ -814,7 +814,7 @@
phaser gain-in gain-out delay decay speed < -s | -t >
Add a phaser to a sound sample. Each triple
- delay/decay/speed gives the delay in millisec-
+ delay/decay/speed gives the delay in millisec�
onds and the decay (relative to gain-in) with a
modulation speed in Hz. The modulation is
either sinodial (-s) or triangular (-t). The
@@ -823,11 +823,11 @@
pick [ -1 | -2 | -3 | -4 | -l | -r ]
Select the left or right channel of a stereo
- sample, or one of four channels in a quadro-
+ sample, or one of four channels in a quadro�
phonic sample. The -l and -r options represent
either the left or right channel. It is
required that you use the -c 1 command line
- option in order to force the output file to con-
+ option in order to force the output file to con�
tain only 1 channel.
pitch shift [ width interpole fade ]
@@ -848,7 +848,7 @@
[ -cutoff # ]
Translate input sampling rate to output sampling
- rate via polyphase interpolation, a DSP algo-
+ rate via polyphase interpolation, a DSP algo�
rithm. This method is slow and uses lots of
@@ -865,11 +865,11 @@
RAM, but gives much better results than rate.
-w < nut / ham > : select either a Nuttal (~90
- dB stopband) or Hamming (~43 dB stopband) win-
+ dB stopband) or Hamming (~43 dB stopband) win�
dow. Default is nut.
- -width long / short / # : specify the (approxi-
- mate) width of the filter. long is 1024 sam-
+ -width long / short / # : specify the (approxi�
+ mate) width of the filter. long is 1024 sam�
ples; short is 128 samples. Alternatively, an
exact number can be used. Default is long. The
short option is not recommended, as it produces
@@ -881,15 +881,15 @@
the resample effect for further information on
Nyquist frequency. If upsampling, then this is
the fraction of the original signal that should
- go through. If downsampling, this is the frac-
+ go through. If downsampling, this is the frac�
tion of the signal left after downsampling.
Default is 0.95. Remember that this is a float.
rate Translate input sampling rate to output sampling
- rate via linear interpolation to the Least Com-
+ 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-
+ 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
@@ -901,13 +901,13 @@
instead use either resample or polyphase. If
you are wondering which rate changing effects to
use, you will want to read a detailed analysis
- of all of them at http://eakaw2.et.tu-dres-
+ of all of them at http://eakaw2.et.tu-dres�
den.de/~wilde/resample/resample.html
resample [ -qs | -q | -ql ] [ rolloff [ beta ] ]
Translate input sampling rate to output sampling
rate via simulated analog filtration. This
- method is slower than rate, but gives much bet-
+ method is slower than rate, but gives much bet�
ter results.
By default, linear interpolation is used, with a
@@ -962,7 +962,7 @@
are indicated above.
The Nyquist frequency is equal to (sample rate /
- 2). Logically, this is because the A/D con-
+ 2). Logically, this is because the A/D con�
verter needs at least 2 samples to detect 1
cycle at the Nyquist frequency. Frequencies
higher then the Nyquist will actually appear as
@@ -972,7 +972,7 @@
these problems.
Similar problems will happen in software when
- reducing the sample rate of an audio file (fre-
+ reducing the sample rate of an audio file (fre�
quencies above the new Nyquist frequency can be
aliased to lower frequencies). Therefore, a
good resample effect will remove all frequency
@@ -979,7 +979,7 @@
information above the new Nyquist frequency.
The rolloff refers to how close to the Nyquist
- frequency this cutoff is, with closer being bet-
+ frequency this cutoff is, with closer being bet�
ter. When increasing the sample rate of an
audio file you would not expect to have any
@@ -995,8 +995,8 @@
frequencies exist that are past the original
- Nyquist frequency. Because of resampling prop-
- erties, it is common to have alaising data cre-
+ Nyquist frequency. Because of resampling prop�
+ erties, it is common to have alaising data cre�
ated that is above the old Nyquist frequency.
In that case the rolloff refers to how close to
the original Nyquist frequency to use a highpass
@@ -1031,7 +1031,7 @@
NOTE: -qs is only slightly slower, but more
accurate for 16-bit or higher precision.
- NOTE: In many cases of up-sampling, no interpo-
+ NOTE: In many cases of up-sampling, no interpo�
lation is needed, as exact filter coefficients
can be computed in a reasonable amount of space.
To be precise, this is done when
@@ -1045,7 +1045,7 @@
is given in milliseconds and its feedback is
depending on the reverb-time in milliseconds.
Each delay should be in the range of half to
- quarter of reverb-time to get a realistic rever-
+ quarter of reverb-time to get a realistic rever�
beration. Gain-out is the volume of the output.
@@ -1069,10 +1069,10 @@
time. A factor of 1.0 means no change, and is
the default. 2.0 doubles speed, thus time
length is cut by a half and pitch is one octave
- higher. 0.5 halves speed thus time length dou-
+ higher. 0.5 halves speed thus time length dou�
bles and pitch is one octave lower.
- split Turn a mono sample into a stereo sample by copy-
+ split Turn a mono sample into a stereo sample by copy�
ing the input channel to the left and right
channels.
@@ -1087,7 +1087,7 @@
will make the sample as loud as possible without
clipping.
- The option -v will print out the "Volume Adjust-
+ The option -v will print out the "Volume Adjust�
ment:" field's value only and return. This
could be of use in scripts to auto convert the
volume.
@@ -1146,6 +1146,19 @@
channel 2's data; creating a stereo file with
both channels containing the same audio data.
+ trim start [ length ]
+ 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 an integer
+ number that tells the exact sample number to
+ start at.
+ 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.
+
vibro speed [ depth ]
Add the world-famous Fender Vibro-Champ sound
effect to a sound sample by using a sine wave as
@@ -1154,35 +1167,22 @@
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 ]
+ 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. 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 vol-
- ume to increase. Beware of clipping audio data
- when the gain is greater then 1.0. A negative
- value performs the same adjustment while also
- 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 change loga-
- rithmically. 0.0 is constant while +6 doubles
- the amplitude.
+ 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 vol�
+ ume to increase. Beware of clipping audio data
- Sox enforces certain effects. If the two files have dif-
- ferent sampling rates, the requested effect must be one of
- polyphase, rate, or resample. If the two files have dif-
- ferent numbers of channels, the avg or other channel mix-
- ing effect must be requested.
-
-
July 24, 2000 18
@@ -1192,11 +1192,27 @@
SoX(1) SoX(1)
+ when the gain is greater then 1.0. A negative
+ value performs the same adjustment while also
+ 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.
+ 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�
+ vent 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
+ 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
+ Please report any bugs found in this version of sox to
Chris Bagwell (cbagwell@sprynet.com)
FILES
@@ -1204,27 +1220,11 @@
play(1), rec(1), soxexam(1)
NOTICES
- The version of Sox that accompanies this manual page is
- support by Chris Bagwell (cbagwell@sprynet.com). Please
+ 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
+ obtain the latest version at the the web site
http://home.sprynet.com/~cbagwell/sox.html
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
--- a/soxexam.txt
+++ b/soxexam.txt
@@ -26,7 +26,7 @@
Most file formats that contain headers can automatically
be read in. When working with headerless file formats
- then a user must manually tell sox the data type and sam-
+ then a user must manually tell sox the data type and sam�
ple rate using command line options.
When working with headerless files (raw files), you may
@@ -113,11 +113,11 @@
or GSM. SoX takes ALL input data types and converts them
to uncompressed 32-bit signed data. It will then convert
this internal version into the requested output format.
- This means unneeded noise can be introduced from decom-
- pressing data and then recompressing. If applying multi-
- ple effects to audio data it is best to save the interme-
- diate data as PCM data. After the final effect is per-
- formed then you can specify it as a compressed output for-
+ This means unneeded noise can be introduced from decom�
+ pressing data and then recompressing. If applying multi�
+ ple effects to audio data it is best to save the interme�
+ diate data as PCM data. After the final effect is per�
+ formed then you can specify it as a compressed output for�
mat. This will keep noise introduction to a minimum.
The following example is to apply various effects to an
@@ -141,7 +141,7 @@
sox thirdfile.wav -a -b finalfile.wav mask
Under a DOS shell, you can convert several audio files to
- an new output format using something similar to the fol-
+ an new output format using something similar to the fol�
lowing command line:
FOR %X IN (*.RAW) DO sox -r 11025 -w -s -t raw $X $X.wav
@@ -232,7 +232,7 @@
concert or ten people are playing the same rhythm with
their drums or funky-groves, then take any other sample.
(Typically, less then four different intruments and no
- synthesizer in the sample is suitable. Likewise, the com-
+ synthesizer in the sample is suitable. Likewise, the com�
bination vocal, drums, bass and guitar.)
Effects:
@@ -245,14 +245,14 @@
not, turn a bit around ant try next, or climb to the next
mountain ).
- However, the time difference between shouting and repeat-
- ing is the delay (time), its loudness is the decay. Multi-
+ However, the time difference between shouting and repeat�
+ ing is the delay (time), its loudness is the decay. Multi�
ple echos can have different delays and decays.
Very popular is using echos to play an instrument with
itself together, like some guitar players ( Brain May from
Queen ) or vocalists are doing. For music samples of more
- than one instrument, echo can be used to add a second sam-
+ than one instrument, echo can be used to add a second sam�
ple shortly after the original one.
This will sound as doubling the number of instruments
@@ -314,7 +314,7 @@
can be applied to other instrument samples too.
It works like the echo effect with a short delay, but the
- delay isn't constant. The delay is varied using a sin-
+ delay isn't constant. The delay is varied using a sin�
odial or triangular modulation. The modulation depth
defines the range the modulated delay is played before or
after the delay. Hence the delayed sound will sound slower
@@ -401,12 +401,12 @@
to small or to many visitors disturb the reflection of
- sound at the walls to make the sound played more monumen-
+ sound at the walls to make the sound played more monumen�
tal. You can try the reverb effect in your bathroom or
garage or sport halls by shouting loud some words. You'll
hear the words reflected from the walls.
- The biggest problem in using the reverb effect is the cor-
+ The biggest problem in using the reverb effect is the cor�
rect setting of the (wall) delays such that the sound is
relistic an doesn't sound like music playing in a tin or
overloaded feedback distroys any illusion of any big hall.
@@ -445,8 +445,8 @@
240.0 280.0 300.0
If you run out of machine power or memory, then stop as
- much applications as possible ( every interupt will con-
- sume a lot of cpu time which for bigger halls is abso-
+ much applications as possible ( every interupt will con�
+ sume a lot of cpu time which for bigger halls is abso�
lutely neccessary ).
Phaser
@@ -466,7 +466,7 @@
SoX(1) SoX(1)
- effects ( simply change the effect name ). The delay mod-
+ effects ( simply change the effect name ). The delay mod�
ulation can be done sinodial or triangular, preferable is
the later one for multiple instruments playing. For single
instrument sounds the sinodial phaser effect will give a
@@ -500,10 +500,10 @@
More effects ( to do ! )
- There are a lot of effects around like noise gates, com-
+ There are a lot of effects around like noise gates, com�
pressors, waw-waw, stereo effects and so on. They should
- be implemented making SOX to be more useful in sound mix-
- ing technics coming together with a great varity of dif-
+ be implemented making SOX to be more useful in sound mix�
+ ing technics coming together with a great varity of dif�
ferent sound effects.
Combining effects be using then in parallel or sequel on
@@ -510,7 +510,7 @@
different channels needs some easy mechanism which is
real-time stable.
- Really missing, is the changing of the parameters, start-
+ Really missing, is the changing of the parameters, start�
ing and stoping of effects while playing samples in real-
time!