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rtpmux: Improve documentation

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Add an example pipeline, and try to explain a bit more what it does.
This commit is contained in:
Olivier Crête 2010-09-30 16:07:29 -04:00 committed by Tim-Philipp Müller
parent ca27a279ba
commit 9674d5cc23
2 changed files with 25 additions and 1 deletions
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4
gst/rtpmanager/gstrtpdtmfmux.c
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@ -33,6 +33,10 @@
* stream. It does exactly what it's parent (#rtpmux) does, except
* that it prevent buffers coming over a regular sink_%%d pad from going through
* for the duration of buffers that came in a priority_sink_%%d pad.
*
* This is especially useful if a discontinuous source like dtmfsrc or
* rtpdtmfsrc are connected to the priority sink pads. This way, the generated
* DTMF signal can replace the recorded audio while the tone is being sent.
*/
#ifdef HAVE_CONFIG_H

22
gst/rtpmanager/gstrtpmux.c
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@ -2,8 +2,10 @@
*
* gstrtpmux.c:
*
* Copyright (C) <2007> Nokia Corporation.
* Copyright (C) <2007-2010> Nokia Corporation.
* Contact: Zeeshan Ali <zeeshan.ali@nokia.com>
* Copyright (C) <2007-2010> Collabora Ltd
* Contact: Olivier Crete <olivier.crete@collabora.co.uk>
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2000,2005 Wim Taymans <wim@fluendo.com>
*
@ -25,9 +27,27 @@
/**
* SECTION:element-rtpmux
* @see_also: rtpdtmfmux
*
* The rtp muxer takes multiple RTP streams having the same clock-rate and
* muxes into a single stream with a single SSRC.
*
* <refsect2>
* <title>Example pipelines</title>
* |[
* gst-launch rtpmux name=mux ! udpsink host=127.0.0.1 port=8888 \
* alsasrc ! alawenc ! rtppcmapay ! \
* application/x-rtp, payload=8, rate=8000 ! mux.sink_0 \
* audiotestsrc is-live=1 ! \
* mulawenc ! rtppcmupay ! \
* application/x-rtp, payload=0, rate=8000 ! mux.sink_1
* ]|
* In this example, an audio stream is captured from ALSA and another is
* generated, both are encoded into different payload types and muxed together
* so they can be sent on the same port.
* </refsect2>
*
* Last reviewed on 2010-09-30 (0.10.21)
*/
#ifdef HAVE_CONFIG_H