sox.txt

visual c++编写关于声音分析的 傅立叶变换．超牛

       bandpass frequency bandwidth
		 Butterworth bandpass filter. Description coming soon!

       bandreject frequency bandwidth
		 Butterworth bandreject filter.	 Description coming soon!

       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 milliseconds and
		 the decay (relative to gain-in) with a modulation speed in Hz
		 using	depth in milliseconds.	The modulation is either sinu-
		 soidal (-s) or triangular (-t).  Gain-out is  the  volume  of
		 the output.

       compand attack1,decay1[,attack2,decay2...]

	       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
		 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  separately  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 the maximum possible  signal  amplitude.   The
		 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
		 -inf,-inf  and 0,0 are assumed; the latter may be overridden,
		 but the former may not.

		 The third (optional) parameter is a post-processing  gain  in
		 dB  which  is	applied after the compression 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 initially,  so	 that,
		 for example, a very large gain is not applied to initial sig-
		 nal 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 compander gain properly adjusts
		 itself.

		 The  fifth  (optional)	 parameter is a delay in seconds.  The
		 input signal is analyzed immediately to control  the  compan-
		 der,  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 oper-
		 ate 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 sampling rate.

       dcshift shift [ limitergain ]
		 DC Shift the audio data, with basic linear amplitude formula.
		 This is most useful if your audio data tends to not  be  cen-
		 tered	around	a value of 0.  Shifting it back will allow you
		 to get the most volume	 adjustments  without  clipping	 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 response of pre-emphasized
		 recordings is rectified.  The filtering  is  defined  in  the
		 standard document ISO 908.

       earwax	 Makes	sound  easier to listen to on headphones.  Adds audio-
		 cues to samples in audio cd format so that when  listened  to
		 on headphones the stereo image is moved from inside your head
		 (standard for headphones) to outside and in front of the lis-
		 tener (standard for speakers). See
		 www.geocities.com/beinges for a full explanation.

       echo gain-in gain-out delay decay [ delay decay ... ]
		 Add  echoing  to a sound sample.  Each delay/decay part gives
		 the delay in milliseconds and the decay (relative to gain-in)
		 of that echo.	Gain-out is the volume of the output.

       echos gain-in gain-out delay decay [ delay decay ... ]
		 Add  a sequence of echos to a sound sample.  Each delay/decay
		 part gives the delay in milliseconds and the decay  (relative
		 to gain-in) of that echo.  Gain-out is the volume of the out-
		 put.

       fade [ type ] fade-in-length

	    [ stop-time [ fade-out-length ] ]
		 Add a fade effect to the beginning, end, or both of the audio
		 data.

		 For fade-ins, this starts from the first sample and ramps the
		 volume of the audio from 0 to full volume over fade-in-length
		 seconds.  Specify 0 seconds if no fade-in is wanted.

		 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 fade-out-length seconds before the stop-time.  If
		 fade-out-length is not specified, it  defaults	 to  the  same
		 value	as  fade-in-length.   No  fade-out is performed if the
		 stop-time is not specified.
		 All times can be specified in either periods of time or  sam-
		 ple   counts.	  To  specify  time  periods  use  the	format
		 hh:mm:ss.frac format.	To specify using sample counts,	 spec-
		 ify  the  number  of samples and append the letter ’s’ to the
		 sample count (for example 8000s).
		 An optional type can be specified to change the type of enve-
		 lope.	 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.

       filter [ low ]-[ high ] [ window-len [ beta ] ]
		 Apply	a  Sinc-windowed lowpass, highpass, or bandpass filter
		 of given window length to the signal.	low refers to the fre-
		 quency of the lower 6dB corner of the filter.	high refers to
		 the frequency of the upper 6dB corner of the filter.

		 A lowpass filter is obtained by leaving low  unspecified,  or
		 0.   A	 highpass  filter is obtained by leaving high unspeci-
		 fied, or 0, or greater than or	 equal	to  the	 Nyquist  fre-
		 quency.

		 The window-len, if unspecified, defaults to 128.  Longer win-
		 dows give a sharper cutoff, smaller windows  a	 more  gradual
		 cutoff.

		 The  beta,  if	 unspecified,  defaults to 16.	This selects a
		 Kaiser window.	 You can select a Nuttall window by specifying
		 anything  <=  2.0  here.   For	 more discussion of beta, look
		 under the resample effect.


       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 milliseconds and the
		 decay (relative to gain-in) with a modulation	speed  in  Hz.
		 The  modulation  is  either sinodial (-s) or triangular (-t).
		 Gain-out is the volume of the output.

       highp frequency
		 Apply a single pole recursive	high-pass  filter.   The  fre-
		 quency response drops logarithmically with I frequency in the
		 middle of the drop.  The slope of the filter is quite gentle.
		 See filter for a highpass effect with sharper cutoff.

       highpass frequency
		 Butterworth highpass filter.  Description coming soon!

       lowp frequency
		 Apply a single pole recursive low-pass filter.	 The frequency
		 response drops logarithmically with frequency in  the	middle
		 of  the  drop.	 The slope of the filter is quite gentle.  See
		 filter for a lowpass effect with sharper cutoff.

       lowpass frequency
		 Butterworth lowpass filter.  Description coming soon!

       mask	 Add "masking noise" to signal.	 This effect deliberately adds
		 white noise to a sound in order to mask quantization effects,
		 created by the process of  playing  a	sound  digitally.   It
		 tends	to  mask buzzing voices, for example.  It adds 1/2 bit
		 of noise to the sound file at the output bit depth.

       mcompand "attack1,decay1[,attack2,decay2...]

		in-dB1,out-dB1[,in-dB2,out-dB2...]

		[gain [initial-volume [delay ] ] ]" xover_freq

		 Multi-band compander is similar to the single band  compander
		 but  the  audio  file is first divided up into bands and then
		 the compander is ran on each band.  See  the  compand	effect
		 for definition of its options.	 Compand options are specified
		 between double quotes and the crossover  frequency  for  that
		 band  is  specefied  seperately  with xover_fre.  This can be
		 repeated multiple times to create multiple bands.

       noiseprof [profile-file]

       noisered profile-file [threshold]
		 Noise reduction filter with profiling. This filter is	moder-
		 ately	effective at removing consistent background noise such
		 as hiss or hum. To use it, first run the noiseprof effect  on
		 a section of silence (that is, a section which contains noth-
		 ing but noise). The noiseprof effect will print a noise  pro-
		 file  to  profile-fire-fR, or to stdout if no profile-file is
		 specified.  If there is sound output on stdout then the  pro-
		 file will next be directed to stderr.

		 To actually remove the noise, run SoX again with the noisered
		 filter. The filter needs one  argument,  profile-file,	 which
		 contains  the	noise profile from noiseprof. thershold speci-
		 fies how much noise should be removed, and may be  between  0
		 and  1	 with a default of 0.5. Higher values will remove more
		 noise but present a greater  possibility  of  distorting  the
		 desired  audio	 signal.   Experiment with different threshold
		 values to find the optimal one for your sample.

       pan direction
		 Pan the sound of an audio file from one channel  to  another.
		 This  is done by changing the volume of the input channels so
		 that it fades out on one channel and fades-in on another.  If
		 the  number of input channels is different then the number of
		 output channels then this effect tries to intelligently  han-
		 dle  this.  For instance, if the input contains 1 channel and
		 the output contains 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 will start the pan effect without totally
		 muting the opposite channel.

       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 milliseconds and the
		 decay (relative to gain-in) with a modulation	speed  in  Hz.
		 The  modulation  is  either sinodial (-s) or triangular (-t).
		 The decay should be less than 0.5 to avoid  feedback.	 Gain-
		 out is the volume of the output.

       pick [ -1 | -2 | -3 | -4 | -l | -r | -f | -b ]
		 Pick  a subset of channels to be copied into the output file.
		 This effect is just an alias of the "avg" effect but is  left
		 here for historical reasons.

       pitch shift [ width interpole fade ]
		 Change	 the  pitch  of file without affecting its duration by
		 cross-fading shifted samples.	shift is given in cents. Use a
		 positive value to shift to treble, negative value to shift to
		 bass.	Default shift is 0.  width of window is in ms. Default
		 width	is  20ms.  Try	30ms to lower pitch, and 10ms to raise
		 pitch.	 interpole option, can be "cubic" or "linear". Default
		 is  "cubic".  The fade option, can be "cos", "hamming", "lin-
		 ear" or "trapezoid".  Default is "cos".

       polyphase [ -w < nut / ham > ]

		 [  -width <  long  / short  / # > ]

		 [ -cutoff #  ]
		 Translate input sampling rate to  output  sampling  rate  via
		 polyphase  interpolation,  a  DSP  algorithm.	This method is
		 slow and uses lots of RAM, but gives much better results than
		 rate.

		 -w  <	nut / ham > : select either a Nuttal (~90 dB stopband)
		 or Hamming (~43 dB stopband) window.  Default is nut.

		 -width long / short / # : specify the (approximate) width  of
		 the  filter.	long  is  1024	samples; short is 128 samples.
		 Alternatively, an exact number can be used.  Default is long.
		 The  short  option  is	 not  recommended, as it produces poor
		 quality results.

		 -cutoff # : specify the filter cutoff frequency in  terms  of
		 fraction  of  frequency  bandwidth,  also know as the Nyquist
		 frequency.  Please see the resample effect for further infor-
		 mation on Nyquist frequency.  If upsampling, then this is the
		 fraction of the original signal that should go	 through.   If
		 downsampling,	this  is the fraction 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 Common Multiple of the  two
		 sampling  rates.  This is 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 orig-
		 inal sound will be shifted  upwards  and  duplicated  faintly
		 when up-translating by a multiple.

		 Lerp-ing  is  acceptable  for cheap 8-bit sound hardware, but
		 for CD-quality sound you should 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-dresden.de/~wilde/resam-
		 ple/resample.html

       repeat count
		 Repeats the audio data count times.  Requires disk  space  to
		 store the data to be repeated.

       resample [ -qs | -q | -ql ] [ rolloff [ beta ] ]