raw audio stream

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raw audio stream

Stanisław Pitucha
Hi,
I'm completely new to gstreamer... could someone tell me how to
configure the pipeline to get audio from a specified (already open) fd
assuming it doesn't have any header and is a raw 8kHz 16 bit signed 1
channel stream?
I know I should use the fdsrc, but how to force a format afterwards?

From what I understood so far, I'll need:
fdsrc fd=3 ! [XXX] ! audioconvert ! audioresample ! vader name=vad
auto-threshold=true ....

What goes in the [XXX] place?

Thanks for any help

--
KTHXBYE,

Stanisław Pitucha, Gradwell Voip Engineer

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Re: raw audio stream

Thiago Sousa Santos-2


2009/10/17 Stanisław Pitucha <[hidden email]>
Hi,
I'm completely new to gstreamer... could someone tell me how to
configure the pipeline to get audio from a specified (already open) fd
assuming it doesn't have any header and is a raw 8kHz 16 bit signed 1
channel stream?
I know I should use the fdsrc, but how to force a format afterwards?

>From what I understood so far, I'll need:
fdsrc fd=3 ! [XXX] ! audioconvert ! audioresample ! vader name=vad
auto-threshold=true ....

What goes in the [XXX] place?

You need a capsfilter and set its "caps" property. In gst-launch, you can just put the caps under ""  and it will work.

Take a look at gst-inspect audioconvert and you'll see (at its pad templates) plenty of raw audio caps examples.
 

Thanks for any help

--
KTHXBYE,

Stanisław Pitucha, Gradwell Voip Engineer

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--
Thiago Sousa Santos

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Re: raw audio stream

Tim-Philipp Müller-2
In reply to this post by Stanisław Pitucha
On Sun, 2009-10-18 at 00:29 +0100, Stanisław Pitucha wrote:

> I'm completely new to gstreamer... could someone tell me how to
> configure the pipeline to get audio from a specified (already open) fd
> assuming it doesn't have any header and is a raw 8kHz 16 bit signed 1
> channel stream?
> I know I should use the fdsrc, but how to force a format afterwards?
>
> From what I understood so far, I'll need:
> fdsrc fd=3 ! [XXX] ! audioconvert ! audioresample ! vader name=vad
> auto-threshold=true ....
>
> What goes in the [XXX] place?

The audioparse element from gst-plugins-bad should do the job here (and
will also ensure correct chunking/buffer sizes).

Cheers
 -Tim



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