shithub: sox

ref: 226fe1c2a6758de617c080ff740427213ec212be
dir: /TODO/

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People are encouraged to pick some of these and implement it. Send all
patches to cbagwell@users.sourceforge.net, or post them on the patches
tracker at http://sf.net/projects/sox/.

SOX includes skeleton format files to assist you in supporting new 
formats, sound effect loops, and special-purpose programs.
The full skeleton format, skel.c, helps you write a driver 
for a new format which has data structures.  skeleff.c is
a starting point for writing a sound effect loop.  Sox.c is
a good starting point for new programs.  (Someone finally
did this and told me what was wrong...)

In handlers.c, note that many formats set up the header and then
use the raw driver for reading and writing.  

  o Update all read routines to be like WAV handler and only
    return data in multiples of sample_size*channels.  This
    will prevent corrupt files from causing sox to go into
    an infinite loop trying to read just a couple more bytes.

  o Make "dither" an alias of "mask" since that's how most
    people refer to this process.

  o Make "mix" an alias of "avg" since that's closer to what it
    really is.  For 2->2 mixes, make a 2 option L->L and R->R
    shortcut.  Similar 4 option for 4->4.

  o In avg effect, going from 1->2 should have a 2 number shortcut
    so that you don't have to put 100% in each channel.  Similar
    for 2->4.

  o Modify noiseprof and noisered so that they can read
    and write from stdin/out.  This would allow to run
    them in scripts without intermediate files required.

  o Several effects have to process a "window" of data at
    one time and sox doesn't always pass that amount in.
    Make some common routines to buffer up a window of data
    instead of requiring each effect to reimplement this.

  o Add an interface to libst so that applications can query
    what size/encoding/rate/channels a given format handles.
    Then applications could modify parameters before an error
    occurs.  This would allow stripping out override logic
    in various formats.

  o Effects engine should not have a global "flow" period
    followed by a global "drain" period.  Each effect should
    track which it needs to invoke.  Until that's done you
    won't be able to stack two effects that both make use
    of drain (ie. can't run the "reverse" effect twice
    on the same command line and have a good output file).

  o Add MP3 support to WAV handler. Can copy most work from mp3.c.

  o Implement speex (http://www.speex.org).
  
  o Modify .dat handler to support stereo data... Probably should remove
    time offset column since it's not real useful (could be added by
    a simple post-processor).  Also, should store information in
    header related to original data so that it can be converted back
    to similar percision easier.  Another idea is to normalize the
    data relative to original percision to reduce value ranges
    (might result in a smaller text file since percision will
    be smaller).

  o Add command line options that specify a specific endian
    (as apposed to -x which just reverses it from current
    machines endian).  This would be useful for cdr.c
    since some formats like to keep the data in little-endien
    (CDDA?).

  o Add a FIFO/buffer to audio drivers to prevent losing audio data.
    Perhaps mpg123 could be used as an example.

  o Lots of effects are using "double" to prevent overflow of
    st_sample_t math.  With ststdint.h we now have a portable way
    of detecting uint64_t...  Use that when possible to speed up
    math (won't have to convert back and forth from int to float).

  o WAV handler is not using word alignment for at least the
    main data chunk (expect for GSM).

  o Keep sox from using "fail" on errors.  Sox was supposed to be
    a sound library called "ST" but libraries shouldn't exit a program,
    they should return error codes for users to handle.  Initial support
    for this has been added.  Needs to be completed.

  o SJB: [11/25/99] (Re. resample effect): Note that upsampling
    usually doesn't require interpolation, therefore is faster and
    more accurate than downsampling. Downsampling by an integer factor
    is also simple, since it just involves decimation if the input is
    already lowpass-filtered to the output Nyquist freqency. Get the
    idea? :) Also, the original algorithm has apparently been updated,
    and might usefully be updated in SoX too.

  o WAV handler needs to handle (ignore if nothing else) non-audio
    chunks (except LIST which is already handled), especially at the
    end of the file.

  o Effects option-parsers should fail if extraneous characters or arguments
    are present. e.g. vol 3mW, speed 2 3, etc. should all fail.