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soxexam



CONVERSIONS

       Introduction

       In  general,  SoX will attempt to take an input sound file
       format and convert it into a new file format using a simi­
       lar  data  type  and sample rate.  For instance, "sox mon­
       key.au monkey.wav" would try and convert the  mono  8000Hz
       u-law  sample  .au file that comes with SoX to a 8000Hz u-
       law .wav file.

       If an output format doesn't support the same data type  as
       the  input  file  then SoX will generally select a default
       data type to save it in.  You  can  override  the  default
       data  type  selection by using command line options.  This
       is also useful for producing an output file with higher or
       lower precision data and/or sample rate.

       Most  file  formats that contain headers can automatically
       be read in.  When working with  header-less  file  formats
       then  a user must manually tell SoX the data type and sam­
       ple rate using command line options.

       When working with header-less files (raw files),  you  may
       take  advantage of the pseudo-file types of .ub, .uw, .sb,
       .sw, .ul, and .sl.  By  using  these  extensions  on  your
       filenames  you  will not have to specify the corresponding
       options on the command line.

       Precision

       The following data types and formats can be represented by
       their  total  uncompressed bit precision.  When converting
       from one data type to another care must be taken to insure
       it  has  an  equal  or greater precision.  If not then the
       audio quality will be degraded.  This is not always a  bad
       thing  when  your  working with things such as voice audio
       and are concerned about disk space  or  bandwidth  of  the
       audio data.

               Data Format    Precision
               ___________    _________
               unsigned byte    8-bit
               signed byte      8-bit
               u-law           14-bit
               A-law           13-bit
               unsigned word   16-bit
               signed word     16-bit
               ADPCM           16-bit
               GSM             16-bit
               unsigned long   32-bit
               signed long     32-bit
               ___________    _________

         sox filename.aiff filename.wav

       To  convert  from mono raw 8000 Hz 8-bit unsigned PCM data
       to a WAV file:

         sox -r 8000 -u -b -c 1 filename.raw filename.wav

       SoX may even be used to convert  sample  rates.   Downcon­
       verting  will  reduce  the bandwidth of a sample, but will
       reduce storage space on your disk.  All  such  conversions
       are  lossy  and  will  introduce  some  noise.  You should
       really pass your sample through a low pass filter prior to
       downconverting  as  this will prevent alias signals (which
       would sound like additional noise).  For example  to  con­
       vert from a sample recorded at 11025 Hz to a u-law file at
       8000 Hz sample rate:

         sox infile.wav -t au -r 8000 -U -b -c 1 outputfile.au

       To add a low-pass filter (note use of stdout for output of
       the first stage and stdin for input on the second stage):

         sox infile.wav -t raw -s -w -c 1 - lowpass 3700  |
           sox  -t  raw -r 11025 -s -w -c 1 - -t au -r 8000 -U -b
       -c 1 ofile.au

       If you hear some clicks and pops when converting to  u-law
       or  A-law,  reduce  the output level slightly, for example
       this will decrease it by 20%:

         sox infile.wav -t au -r 8000 -U -b -c 1  -v  .8  output­
       file.au

       SoX is great to use along with other command line programs
       by passing data between the programs using pipelines.  The
       most  common example is to use mpg123 to convert mp3 files
       in to wav files.  The following command line will do this:

         mpg123 -b 10000 -s filename.mp3 | sox -t raw -r 44100 -s
       -w -c 2 - filename.wav

       When working with totally  unknown  audio  data  then  the
       "auto"  file  format  may be of use.  It attempts to guess
       what the file type is and then you  may  save  it  into  a
       known audio format.

         sox -V -t auto filename.snd filename.wav

       It  is  important  to  understand how the internals of SoX
       work with compressed audio including u-law, A-law,  ADPCM,
         sox firstfile.wav -r 44100 -s -w secondfile.wav
         sox secondfile.wav thirdfile.wav swap
         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
       lowing command line:

         FOR %X IN (*.RAW) DO sox -r 11025 -w -s -t raw $X $X.wav


EFFECTS

       Special     thanks     goes     to     Juergen     Mueller
       (jmeuller@uia.au.ac.be) for this write up on effects.

       Introduction:

       The core problem is that you need some experience in using
       effects in order to say "that any old  sound  file  sounds
       with  effects  absolutely hip". There isn't any rule-based
       system which tell you  the  correct  setting  of  all  the
       parameters for every effect.  But after some time you will
       become an expert in using effects.

       Here are some examples which can be used  with  any  music
       sample.   (For  a sample where only a single instrument is
       playing, extreme parameter  setting  may  make  well-known
       "typically"  or  "classical"  sounds. Likewise, for drums,
       vocals or guitars.)

       Single effects will be explained and some given  parameter
       settings that can be used to understand the theory by lis­
       tening to the sound file with the added effect.

       Using multiple effects in parallel or in series can result
       either  in  a  very  nice  sound or (mostly) in a dramatic
       overloading in variations of sounds such that your ear may
       follow the sound but you will feel unsatisfied. Hence, for
       the first time using effects try to compose them as  mini­
       mally  as  possible.  We  don't  regard the composition of
       effects in the examples because too many combinations  are
       possible and you really need a very fast machine and a lot
       of memory to play them in real-time.

       However, real-time playing of sounds will greatly speed up
       learning  and/or  tuning  the  parameter settings for your
       sounds in order to get that "perfect" effect.

       Basically, we will use the "play" front-end of  SoX  since
       it is easier to listen sounds coming out of the speaker or
       earphone instead of  looking  at  cryptic  data  in  sound
       files.

       effect-parameters

       And for date freaks:

             sox file.xxx file.yyy effect-name effect-parameters

       Additional options can be used. However, in this case, for
       real-time playing you'll need a very fast machine.

       Notes:

       I played all examples in real-time on a Pentium  100  with
       32 MB and Linux 2.0.30 using a self-recorded sample ( 3:15
       min long in "wav" format with 44.1  kHz  sample  rate  and
       stereo 16 bit ).  The sample should not contain any of the
       effects. However, if you take any  recording  of  a  sound
       track  from radio or tape or CD, and it sounds like a live
       concert or ten people are playing  the  same  rhythm  with
       their  drums or funky-grooves, then take any other sample.
       (Typically, less then four different  instruments  and  no
       synthesizer  in the sample is suitable. Likewise, the com­
       bination vocal, drums, bass and guitar.)

       Effects:

       Echo

       An echo effect can be naturally found  in  the  mountains,
       standing  somewhere  on  a  mountain and shouting a single
       word will result in one or more repetitions  of  the  word
       (if  not, turn a bit around and try again, 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
       ple echos can have different delays and decays.

       It is very popular to use 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
       ple shortly after the original one.

       This will sound as if  you  are  doubling  the  number  of
       instruments playing in the same sample:

             play file.xxx echo 0.8 0.88 60.0 0.4

       If  the  delay is very short, then it sound like a (metal­
       lic) robot playing music:

             play file.xxx echo 0.8 0.88 6.0 0.4

       input  and  the  first  echos, the third the input and the
       first and the second echos, ... and so on.  Care should be
       taken  using many echos (see introduction); a single echos
       has the same effect as a single echo.

       The sample will be bounced twice in symmetric echos:

             play file.xxx echos 0.8 0.7 700.0 0.25 700.0 0.3

       The sample will be bounced twice in asymmetric echos:

             play file.xxx echos 0.8 0.7 700.0 0.25 900.0 0.3

       The sample will sound as if played in a garage:

             play file.xxx echos 0.8 0.7 40.0 0.25 63.0 0.3

       Chorus

       The chorus effect has its name because it  will  often  be
       used  to  make  a single vocal sound like a chorus. But it
       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 sinu­
       soidal 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
       or  faster,  that  is  the  delayed sound tuned around the
       original one, like in a chorus where some vocals are a bit
       out of tune.

       The typical delay is around 40ms to 60ms, the speed of the
       modulation is best near 0.25Hz and  the  modulation  depth
       around 2ms.

       A single delay will make the sample more overloaded:

             play file.xxx chorus 0.7 0.9 55.0 0.4 0.25 2.0 -t

       Two delays of the original samples sound like this:

              play  file.xxx  chorus 0.6 0.9 50.0 0.4 0.25 2.0 -t
       60.0 0.32 0.4 1.3 -s

       A big chorus of the sample is (three additional samples):

             play file.xxx chorus 0.5 0.9 50.0 0.4  0.25  2.0  -t
       60.0 0.32 0.4 2.3 -t        40.0 0.3 0.3 1.3 -s

       Flanger

             play file.xxx flanger 0.6 0.87 3.0 0.9 0.5 -s

       listen  carefully between the difference of sinusoidal and
       triangular modulation:

             play file.xxx flanger 0.6 0.87 3.0 0.9 0.5 -t

       If the decay is a bit lower, than the effect  sounds  more
       popular:

             play file.xxx flanger 0.8 0.88 3.0 0.4 0.5 -t

       The drunken loudspeaker system:

             play file.xxx flanger 0.9 0.9 4.0 0.23 1.3 -s

       Reverb

       The reverb effect is often used in audience hall which are
       to small or contain too many many visitors  which  disturb
       (dampen)  the  reflection  of  sound at the walls.  Reverb
       will make the sound be perceived as if it were in a  large
       hall.   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
       rect setting of the (wall) delays such that the  sound  is
       realistic  and  doesn't  sound like music playing in a tin
       can or has overloaded feedback which destroys any illusion
       of  playing  in  a big hall.  To help you obtain realistic
       reverb effects, you  should  decide  first  how  long  the
       reverb should take place until it is not loud enough to be
       registered by your ears. This is be done  by  varying  the
       reverb  time "t".  To simulate small halls, use 200ms.  To
       simulate large halls, use 1000ms.  Clearly, the  walls  of
       such a hall aren't far away, so you should define its set­
       ting be given every wall its delay time.  However, if  the
       wall  is  to  far away for the reverb time, you won't hear
       the reverb, so the nearest wall  will  be  best  at  "t/4"
       delay  and  the  farthest at "t/2". You can try other dis­
       tances as well, but it won't sound  very  realistic.   The
       walls  shouldn't stand to close to each other and not in a
       multiple integer distance to each other (  so  avoid  wall
       like:  200.0  and 202.0, or something like 100.0 and 200.0
       ).

       Since audience halls do have a lot of walls, we will start
       designing one beginning with one wall:

             play file.xxx reverb 1.0 600.0 180.0

       If  you  run  out of machine power or memory, then stop as
       many applications as possible (every interrupt  will  con­
       sume  a  lot  of  CPU time which for bigger halls is abso­
       lutely necessary).

       Phaser

       The phaser effect is like the flanger effect, but it  uses
       a  reverb  instead  of  an  echo  and does phase shifting.
       You'll hear the difference in the examples comparing  both
       effects  (simply change the effect name).  The delay modu­
       lation can be sinusoidal or triangular, preferable is  the
       later  for  multiple  instruments.  For  single instrument
       sounds, the sinusoidal phaser effect will give  a  sharper
       phasing  effect.   The  decay shouldn't be to close to 1.0
       which will cause dramatic feedback.  A good range is about
       0.5 to 0.1 for the decay.

       We will take a parameter setting as for the flanger before
       (gain-out is lower since feedback  can  raise  the  output
       dramatically):

             play file.xxx phaser 0.8 0.74 3.0 0.4 0.5 -t

       The drunken loudspeaker system (now less alcohol):

             play file.xxx phaser 0.9 0.85 4.0 0.23 1.3 -s

       A popular sound of the sample is as follows:

             play file.xxx phaser 0.89 0.85 1.0 0.24 2.0 -t

       The sample sounds if ten springs are in your ears:

             play file.xxx phaser 0.6 0.66 3.0 0.6 2.0 -t

       Compander

       The  compander effect allows the dynamic range of a signal
       to be compressed or expanded.  For  most  situations,  the
       attack  time (response to the music getting louder) should
       be shorter than the decay time because our ears  are  more
       sensitive  to  suddenly  loud  music than to suddenly soft
       music.

       For example, suppose you are listening to  Strauss'  "Also
       Sprach  Zarathustra" in a noisy environment such as a car.
       If you turn up the volume enough to hear the soft passages
       over  the  road noise, the loud sections will be too loud.
       You could try this:

                   play       file.xxx       compand        0.3,1

       fine for a clip that starts with a bit of silence, and the
       delay of 0.2 has the effect of causing  the  compander  to
       react a bit more quickly to sudden volume changes.

       Changing the Rate of Playback

       You  can  use stretch to change the rate of playback of an
       audio sample while preserving the pitch.  For  example  to
       play at 1/2 the speed:

             play file.wav stretch 2

       To play a file at twice the speed:

             play file.wav stretch .5

       Other  related  options are "speed" to change the speed of
       play (and changing the pitch accordingly), and  pitch,  to
       alter  the pitch of a sample.  For example to speed a sam­
       ple so it plays in 1/2 the time (for  those  Mickey  Mouse
       voices):

             play file.wav speed 2

       To raise the pitch of a sample 1 while note (100 cents):

             play file.wav pitch 100

       Other  effects (copy, rate, avg, stat, vibro, lowp, highp,
       band, reverb)

       The other effects are simple to use. However, an "easy  to
       use manual" should be given here.

       More effects (to do !)

       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  more useful in sound mixing
       techniques coming together with a great variety of differ­
       ent sound effects.

       Combining effects by using them in parallel or serially on
       different channels needs some easy mechanism which is sta­
       ble for use in real-time.

       Really  missing are the the changing of the parameters and
       starting/stopping of  effects  while  playing  samples  in
       real-time!


                        December 11, 2001                  SoX(1)

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