sox.txt

visual c++编写关于声音分析的 傅立叶变换．超牛

		 Translate  input  sampling  rate  to output sampling rate via
		 simulated analog filtration.	This  method  is  slower  than
		 rate, but gives much better results.

		 By default, linear interpolation is used, with a window width
		 about 45 samples at the lower of the two rate.	 This gives an
		 accuracy  of  about 16 bits, but insufficient stopband rejec-
		 tion in the case that you want to have rolloff	 greater  than
		 about 0.80 of the Nyquist frequency.

		 The  -q*  options  will change the default values for rolloff
		 and beta as well as use  quadratic  interpolation  of	filter
		 coefficients, resulting in about 24 bits precision.  The -qs,
		 -q, or -ql options specify increased accuracy at the cost  of
		 lower execution speed.	 It is optional to specify rolloff and
		 beta parameters when using the -q* options.

		 Following is a table of the  reasonable  defaults  which  are
		 built-in to SoX:

		    Option  Window rolloff beta interpolation
		    ------  ------ ------- ---- -------------
		    (none)    45    0.80    16	   linear
		      -qs     45    0.80    16	  quadratic
		      -q      75    0.875   16	  quadratic
		      -ql    149    0.94    16	  quadratic
		    ------  ------ ------- ---- -------------

		 -qs, -q, or -ql use window lengths of 45, 75, or 149 samples,
		 respectively, at the lower  sample-rate  of  the  two	files.
		 This means progressively sharper stop-band rejection, at pro-
		 portionally slower execution times.

		 rolloff refers to the cut-off frequency of the low pass  fil-
		 ter  and  is  given in terms of the Nyquist frequency for the
		 lower sample rate.  rolloff  therefore	 should	 be  something
		 between  0.0 and 1.0, in practice 0.8-0.95.  The defaults are
		 indicated above.

		 The Nyquist frequency is equal to (sample rate /  2).	 Logi-
		 cally,	 this  is  because  the A/D converter needs at least 2
		 samples to detect 1 cycle at the Nyquist frequency.  Frequen-
		 cies  higher  then  the Nyquist will actually appear as lower
		 frequencies to the A/D	 converter  and	 is  called  aliasing.
		 Normally, A/D converts run the signal through a highpass fil-
		 ter first to avoid these problems.

		 Similar problems will happen in software  when	 reducing  the
		 sample	 rate  of  an  audio  file  (frequencies above the new
		 Nyquist frequency  can	 be  aliased  to  lower	 frequencies).
		 Therefore,  a	good resample effect will remove all frequency
		 information above the new Nyquist frequency.

		 The rolloff refers to how close to the Nyquist frequency this
		 cutoff	 is,  with  closer  being better.  When increasing the
		 sample rate of an audio file you would not expect to have any
		 frequencies  exist  that  are	past the original Nyquist fre-
		 quency.  Because of resampling properties, it	is  common  to
		 have aliasing data created that is above the old Nyquist fre-
		 quency.  In that case the rolloff refers to how close to  the
		 original Nyquist frequency to use a highpass filter to remove
		 this false data, with closer also being better.

		 The beta parameter determines the type of filter window used.
		 Any  value  greater than 2.0 is the beta for a Kaiser window.
		 Beta <= 2.0 selects a Nuttall window.	 If  unspecified,  the
		 default is a Kaiser window with beta 16.

		 In  the  case of Kaiser window (beta > 2.0), lower betas pro-
		 duce a somewhat faster transition from passband to  stopband,
		 at  the  cost	of  noticeable artifacts.  A beta of 16 is the
		 default, beta less than 10 is not recommended.	 If you want a
		 sharper  cutoff,  don’t  use  low beta’s, use a longer sample
		 window.  A Nuttall  window  is	 selected  by  specifying  any
		 ’beta’ <= 2, and the Nuttall window has somewhat steeper cut-
		 off than the default Kaiser window.  You  will	 probably  not
		 need  to  use	the beta parameter at all, unless you are just
		 curious about comparing the effects  of  Nuttall  vs.	Kaiser
		 windows.

		 This  is  the	default effect if the two files have different
		 sampling rates.  Default parameters are, as indicated	above,
		 Kaiser	 window	 of  length  45, rolloff 0.80, beta 16, linear
		 interpolation.

		 NOTE: -qs is only slightly  slower,  but  more	 accurate  for
		 16-bit or higher precision.

		 NOTE:	In  many  cases	 of  up-sampling,  no interpolation is
		 needed, as exact filter coefficients can  be  computed	 in  a
		 reasonable amount of space.  To be precise, this is done when

			    input_rate < output_rate
				       &&
		   output_rate/gcd(input_rate,output_rate) <= 511

       reverb gain-out reverbe-time delay [ delay ... ]
		 Add reverberation to a sound sample.  Each delay 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 reverberation.
		 Gain-out is the volume of the output.

       reverse	 Reverse the sound sample completely.	Included  for  finding
		 Satanic subliminals.

       silence above_periods [ duration threshold[ d | % ]

	       [ below_periods duration

		 threshold[ d | % ]]
		 Removes  silence  from	 the beginning or end of a sound file.
		 Silence is anything below a specified threshold.
		 When trimming silence from the beginning of a sound file, you
		 specify  a  duration  of  audio that is above a given silence
		 threshold before audio data is processed.  You can also spec-
		 ify  the  count of periods of none-silence you want to detect
		 before processing audio data.	Specify a period of 0  if  you
		 do not want to trim data from the front of the sound file.
		 When  optionally  trimming  silence  form  the end of a sound
		 file, you specify the duration of audio that must be below  a
		 given	threshold  before  stopping  to process audio data.  A
		 count of periods that occur below the threshold may  also  be
		 specified.   If  this	options are not specified then data is
		 not trimmed from the end of the audio file.  If below_periods
		 is  negative,	it  is treated as a positive value and is also
		 used to indicate the  effect  should  restart	processing  as
		 specified by the above_periods, making it suitable for remov-
		 ing periods of silence in the middle of a sound file.
		 Duration counts may be in the format of time,	hh:mm:ss.frac,
		 or in the exact count of samples.
		 Threshold may be suffixed with d, or % to indicated the value
		 is in decibels or a percentage of max	value  of  the	sample
		 value.	 A value of ’0%’ will look for total silence.

       speed [ -c ] factor
		 Speed	up  or down the sound, as a magnetic tape with a speed
		 control.  It affects both pitch and 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 doubles and pitch is one
		 octave lower.	If the optional -c parameter is used then  the
		 factor is specified in "cents".

       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
		 option.

		 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
		 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
		 this fact.

		 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-
		 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
		 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
		 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
		 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 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,
		 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 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
		 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
		 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
		 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
		 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
		 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
		 in volume.
		 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
		 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
		 to be limited.

BUGS
       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
       ([e-mail])

FILES
SEE ALSO
       play(1), rec(1), soxexam(1)

NOTICES
       The  version  of	 SoX  that  accompanies this manual page is support by
       Chris Bagwell ([e-mail]).	Please refer any ques-
       tions  regarding it to this address.  You may obtain the latest version
       at the the web site http://sox.sourceforge.net/

AUTHOR
       Chris Bagwell ([e-mail]).

       Updates by Anonymous



			       December 11, 2001			SoX(1)