[Bug 597822] Add removesilence plugin

[GStreamer (bugzilla.gnome.org)]

https://bugzilla.gnome.org/show_bug.cgi?id=597822
  GStreamer | gst-plugins-bad | git

--- Comment #19 from tiagokatcipis at gmail.com 2011-05-31 14:01:00 UTC ---
> But you don't need GST_PLUGIN_BASE_{LIBS,CFLAGS} and GST_BASE_{LIBS,CFLAGS} at
all anyway

ill remove them.

> We have 2011 now :)

The initial proposal was made on 2009, since them little change has been made
on the code, until now of course :-).

> You can use rate=[1, MAX]

It definitely better.

> Also it would be nice to have support for multiple channels, should be easy to add

I think for now mono is fine (i never had the need for multiple channels), if
people starts to use it and there is a need for multichannel ill do it.

> You can use "endianness = BYTE_ORDER"

I just copied the code from another plugin, ill fix it.

> Really try to use GstBaseTransform here

I will, but are you sure i will be able to have the same behaviour using
GstBaseTransform ?

> Now I understand what you're doing here. You want to use gst_audio_buffer_clip() here, it does what you're doing and clips the buffer in other situations too

I will take a look at gst_audio_buffer_clip.

> Better use the GLib types everywhere, gint16 instead of short, etc. In this file and in the vad_private.[ch]

i did it, but i forgot some non GLIB types, i will re review it :-).


> You should have a GstSegment member in your instance structure and use gst_segment_set_newsegment_full() with the values of the event (or the bytes->time transformed values). If you use GstBaseTransform as base class this will already be done for you and you can use the segment member of GstBaseTransform. And you don't need any segment handling at all anymore and none of the gst_pad_set_*function() functions

I'm going to use GstBaseTransform. Thanks a lot for the help Sebastian, I'm
learning a lot at each patch, the next one might take some time but the
important thing is that it is improving.

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