[Sugar-devel] gst-plugins-espeak port to gstreamer 1.0

Aleksey Lim alsroot at sugarlabs.org
Sun Aug 19 03:02:39 EDT 2012


On Sat, Aug 11, 2012 at 10:06:26AM +0200, Daniel Narvaez wrote:
> I managed to fix the issue with the last part not being played and
> posted a patch with the port.

Did you miss to attach new patch.
Any way, I took your initial patch, tuned it and pushed initial port to
master branch.

> I still think returning an arbitrary
> size buffer is wrong and should be fixed at some point.

The original buffer size should not matter, if I got gst docs right,
only `offset` and `size` does matter
http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstBuffer.html#gst-buffer-new-wrapped-full

> On 10 August 2012 20:40, Daniel Narvaez <dwnarvaez at gmail.com> wrote:
> > I got this to work somewhat, it fails to play the last word for some
> > reason though.
> >
> > I'm not convinced the original code is correct. It returns a GstBuffer
> > which is normally bigger than the one requested by the GstBaseSrc
> > create method. And according to the documentation that's not allowed.
> > Though I know very little about gstreamer and nothing about espeak, so
> > I could be very well wrong!
> >
> > On 8 August 2012 22:47, Daniel Narvaez <dwnarvaez at gmail.com> wrote:
> >> Hello,
> >>
> >> I made good progress on porting gst-plugins-espeak to gstreamer 1.0.
> >> Unfortunately I can't quite get it to work. If I save the raw audio
> >> with the filesink it's exactly the same as with gstream 0.10. But I
> >> can't get it to play with
> >>
> >> gst-launch-1.0 espeak text="Hello world" ! autoaudiosink

I compiled gst from git and got broken `autoaudiosink` behaviour. For
example

    gst-launch -v -m audiotestsrc ! audioconvert ! audioresample ! autoaudiosink

fails with the same error as

    gst-launch-1.0 espeak text="Hello world" ! autoaudiosink

So, it seems to gst development code fails. I used another pipeline:

    gst-launch-1.0 espeak text="Hello world" ! wavenc ! filesink location=probe.wav

-- 
Aleksey


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