[gst-devel] RTP audio stream to filesink, issue with packet loss

[Thorsten Brendt]

On Thu, Jan 13, 2011 at 12:16 AM, Tim-Philipp Müller <t.i.m at zen.co.uk> wrote:
> On Wed, 2011-01-12 at 22:21 +0100, Thorsten Brendt wrote:
>
> Hi,
>
>> (snip)
>>
>> Ultimately some element in the pipeline needs to produce
>>  "silence/empty/zero data" if its input buffer is empty.  Is there an
>>   explicit way (helper element/attribute) to do this?
>
> Tried adding an audiorate element?

Do you mean a caps-filter element like this?:

'audio/x-raw-int, width=16, depth=16, rate=8000, endianness=1234,
signed=(boolean)true, channels=1'

The right place to put this seems to be before the encoder element
(wavenc in my case), however in that case the pipeline can't be linked
together.  It works before an autoaudiosink though.

Anyhow, I assume that the audio's sampling rate is known to the
encoder already, as it both given explicitly as a clock-rate attribute
in a previous caps-filter and should be implicitly known by the
mulaw-decoder (plus the audio file "sounds right" in general).