shithub: sox

--- a/sox.1

+++ b/sox.1

@@ -101,7 +101,7 @@

.EX

 	sox -r 8k -u -b 8 -c 1 voice-memo.raw voice-memo.wav

.EE

-converts `raw' (a.k.a. `headerless') audio to a self-descibing file format,

+converts `raw' (a.k.a. `headerless') audio to a self-describing file format,

.EX

 	sox slow.aiff fixed.aiff speed 1.027

.EE

@@ -282,12 +282,12 @@

 to perform the necessary sample rate conversion.  For

 compatibility with old hardware, here, the

 default \fBrate\fR quality level is set to `low'; however, this

-can be changed if desired, by explicitly specifing the \fBrate\fR

+can be changed if desired, by explicitly specifying the \fBrate\fR

 effect with a different quality level, e.g.

.EX

 	play ... rate -m

.EE

-or by setting the environment varible

+or by setting the environment variable

 .B PLAY_RATE_ARG

 to the desired quality option, e.g.

.EX

@@ -350,7 +350,7 @@

.EE

 SoX first decompresses the input MP3 file, then applies the

 .B trim

-effect, and finally creates the output MP3 file by recompressing the

+effect, and finally creates the output MP3 file by re-compressing the

 audio\*mwith a possible reduction in fidelity above that which

 occurred when the input file was created.

 Hence, if what is ultimately desired is lossily compressed audio, it is

@@ -373,7 +373,7 @@

 other effects, when converting one format to another, and even when

 simply playing the audio.

.SP

-Playing an audio file often involves re-sampling, and processing by

+Playing an audio file often involves resampling, and processing by

 analogue components that can introduce a small DC offset and/or

 amplification, all of which can produce distortion if the audio signal

 level was initially too close to the clipping point.

@@ -430,7 +430,7 @@

 same, however, SoX will issue a warning if they are not and some

 channels in the output file will not contain audio from every input

 file.  A mixed audio file cannot be un-mixed (without reference to the

-orignal input files).

+original input files).

.SP

 If the `merge' combining method is selected, then two or

 more input files must be given and will be merged together to form the

@@ -488,12 +488,12 @@

 \(S1/\s-2\(srn\s+2 instead of \(S1/\s-2n\s+2.

 Note that this balancing factor does not guarantee that no clipping will occur,

 however, in many cases, the number of clips will be low and the resultant

-distortion imperceptable.

+distortion imperceptible.

 .SS Output Files

-SoX's default behavior is to take one or more input files and

+SoX's default behaviour is to take one or more input files and

 write them to a single output file.

-This behavior can be changed by specifying the pseudo-effect 'newfile'

+This behaviour can be changed by specifying the pseudo-effect 'newfile'

 within the effects list.  SoX will then enter multiple output mode.

 In multiple output mode, a new file is created when the effects

@@ -503,7 +503,7 @@

 In multiple output mode, a unique number will automatically be appended

 to the end of all filenames.  If the filename has an extension

-then the number is inserted before the extension.  This behavior can

+then the number is inserted before the extension.  This behaviour can

 be customized by placing a %n anywhere in the filename where the

 number should be substituted.  An optional number can be placed after

 the % to indicate a minimum fixed width for the number.

@@ -839,7 +839,7 @@

 Show SoX's version number and exit.

.TP

 \fB\-\-single-threaded\fR

-On some hyper-threading/mult-core architectures,

+On some hyper-threading/multi-core architectures,

 SoX has support for parallel effects channel processing.

 This option can be given to disable parallel processing.

 .IP \fB\-V\fB[\fIlevel\fB]\fP

@@ -1346,7 +1346,7 @@

 parameters may be used to adjust these parameters and thus control the

 smoothness of the changes in pitch.

.SP

-For example, an initial tone is generated, then bent three times, yeilding

+For example, an initial tone is generated, then bent three times, yielding

 four different notes in total:

.EX

 	play -n synth 2.5 sin 667 gain 1 \\

@@ -1524,7 +1524,7 @@

.TP

 \fBdcshift \fIshift\fR [\fIlimitergain\fR]

 DC Shift the audio, with basic linear amplitude formula.

-This is most useful if your audio tends to not be centered around

+This is most useful if your audio tends to not be centred around

 a value of 0.  Shifting it back will allow you to get the most volume

 adjustments without clipping.

.SP

@@ -1605,7 +1605,7 @@

 	  %-2 delay 0 .05 .1 .15 .2 .25 remix - fade 0 4 .1 norm -1

.EE

.TP

-\fBdither\fR [\fB\-r\fR|\fB\-t\fR] [\fB\-s\fR|\fB\-f \fIfilter\fR] [\fIdepth\fR]

+\fBdither\fR [\fB\-R\fR] [\fB\-r\fR\^|\^\fB\-t\fR] [\fB\-s\fR\^|\^\fB\-f \fIfilter\fR] [\fIdepth\fR]

 Apply dithering to the audio.

 Dithering deliberately adds a small amount of noise to the signal

 in order to mask audible quantization effects that

@@ -1625,7 +1625,7 @@

 the following list: lipshitz, f-weighted, modified-e-weighted,

 improved-e-weighted, gesemann, shibata, low-shibata, high-shibata.

 Note that most filter types are available only with 44100Hz sample rate.

-The filter types are distiguished by the following properties:

+The filter types are distinguished by the following properties:

 audibility of noise, level of (inaudible, but in some circumstances,

 otherwise problematic) shaped high frequency noise, and

 processing speed.

@@ -1637,6 +1637,12 @@

 of the added white noise, but correspondingly creates residual

 quantization noise, so it should not normally be changed.

.SP

+If

+.B \-R

+option is given, then the pseudo-random number generator used to

+generate the white noise will be `reseeded', i.e. the generated noise

+will be different between invocations.

+.SP

 This effect should not be followed by any other effect that

 affects the audio.

.TP

@@ -1657,7 +1663,7 @@

 effects emulate this behaviour and are often used to help fill

 out the sound of a single instrument or vocal.  The time difference

 between the original signal and the reflection is the `delay' (time),

-and the loudness of the relected signal is the `decay'.  Multiple echoes

+and the loudness of the reflected signal is the `decay'.  Multiple echoes

 can have different delays and decays.

.SP

 Each given

@@ -1848,7 +1854,7 @@

 .B \-e

 option, the levels of the audio channels of a multi-channel file are `equalised', i.e.

 gain is applied to all channels other than that with the highest peak

-level, such that all channels atain the same peak level

+level, such that all channels attain the same peak level

 (but, without also giving

 .BR \-n ,

 the audio is not `normalised').

@@ -1882,7 +1888,7 @@

.SP

The

 .B \-r

-option is used in conjuction with a prior invocation of

+option is used in conjunction with a prior invocation of

 .B gain

 with the

 .B \-h

@@ -1890,7 +1896,7 @@

.SP

The

 .B \-n

-option normalises the audio to 0dB FSD; it is often used in conjuction with a negative

+option normalises the audio to 0dB FSD; it is often used in conjunction with a negative

 .I gain-dB

 to the effect that the audio is normalised to a given level below 0dB.

 For example,

@@ -1979,7 +1985,7 @@

 module can contain more than one plugin) and any other arguments are

 for the control ports of the plugin. Missing arguments are supplied by

 default values if possible. Only plugins with at most one audio input

-and one audio output port can be used.  If found, the environment varible

+and one audio output port can be used.  If found, the environment variable

 LADSPA_PATH will be used as search path for plugins.

.TP

 \fBloudness\fR [\fIgain\fR [\fIreference\fR]]

@@ -2033,7 +2039,7 @@

.EE

 The audio file is played with a simulated FM radio sound (or broadcast

 signal condition if the lowpass filter at the end is skipped).

-Note that the pipeline is set up with US-style 75us preemphasis.

+Note that the pipeline is set up with US-style 75us pre-emphasis.

.SP

 See also

 .B compand

@@ -2165,7 +2171,7 @@

 \fBoverdrive\fR [\fIgain\fR(20) [\fIcolour\fR(20)]]

 Non linear distortion.

 The \fIcolour\fR parameter controls the amount of even harmonic content

-in the overdriven output.

+in the over-driven output.

.TP

 \fBpad\fR { \fIlength\fR[\fB@\fIposition\fR] }

 Pad the audio with silence, at the beginning, the end, or any

@@ -2340,7 +2346,7 @@

 All resamplers use filters that can sometimes create `echo' (a.k.a.

 `ringing') artefacts with transient signals such as those that occur

 with `finger snaps' or other highly percussive sounds.  Such artefacts are

-much more noticable to the human ear if they occur before the transient

+much more noticeable to the human ear if they occur before the transient

 (`pre-echo') than if they occur after it (`post-echo').  Note that

 frequency of any such artefacts is related to the smaller of the

 original and new sampling rates but that if this is at least 44\*d1kHz,

@@ -2365,7 +2371,7 @@

 (greater than 50) are rarely useful.

.SP

 A resampler's band-width setting determines how much of the frequency

-content of the original signal (w.r.t. the orignal sample rate when

+content of the original signal (w.r.t. the original sample rate when

 up-sampling, or the new sample rate when down-sampling) is preserved

 during conversion.  The term `pass-band' is used to refer to all frequencies

 up to the band-width point (e.g. for 44\*d1kHz sampling rate, and a

@@ -2702,11 +2708,11 @@

 given, then \fIfreqHP\fR < \fIfreqLP\fR creates a band-pass filter,

 \fIfreqHP\fR > \fIfreqLP\fR creates a band-reject filter.

.SP

-The default stop-band attenuation of 120dB can be overriden with

+The default stop-band attenuation of 120dB can be overridden with

 \fB\-a\fR; alternatively, the kaiser-window `beta' parameter can be

 given directly with \fB\-b\fR.

.SP

-The default transition band-width of 5% of the total band can be overriden with

+The default transition band-width of 5% of the total band can be overridden with

 \fB\-t\fR; alternatively, the number of filter taps can be

 given directly with \fB\-n\fR.

.SP