[gst-cvs] gst-plugins-bad: rtpmux: Improve documentation

[<te...@ke...>]

Module: gst-plugins-bad
Branch: master
Commit: 51887b934f701497a8842959447c0d8d1610f9e9
URL:    http://cgit.freedesktop.org/gstreamer/gst-plugins-bad/commit/?id=51887b934f701497a8842959447c0d8d1610f9e9

Author: Olivier Crête <oli...@co...>
Date:   Thu Sep 30 16:07:29 2010 -0400

rtpmux: Improve documentation

Add an example pipeline, and try to explain a bit more what it does.

---

 gst/rtpmux/gstrtpdtmfmux.c |    4 ++++
 gst/rtpmux/gstrtpmux.c     |   22 +++++++++++++++++++++-
 2 files changed, 25 insertions(+), 1 deletions(-)

diff --git a/gst/rtpmux/gstrtpdtmfmux.c b/gst/rtpmux/gstrtpdtmfmux.c
index c953910..f62c626 100644
--- a/gst/rtpmux/gstrtpdtmfmux.c
+++ b/gst/rtpmux/gstrtpdtmfmux.c
@@ -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
diff --git a/gst/rtpmux/gstrtpmux.c b/gst/rtpmux/gstrtpmux.c
index 239b6d3..f014400 100644
--- a/gst/rtpmux/gstrtpmux.c
+++ b/gst/rtpmux/gstrtpmux.c
@@ -2,8 +2,10 @@
  *
  * gstrtpmux.c:
  *
- * Copyright (C) <2007> Nokia Corporation.
+ * Copyright (C) <2007-2010> Nokia Corporation.
  *   Contact: Zeeshan Ali <zee...@no...>
+ * Copyright (C) <2007-2010> Collabora Ltd
+ *   Contact: Olivier Crete <oli...@co...>
  * Copyright (C) 1999,2000 Erik Walthinsen <om...@cs...>
  *               2000,2005 Wim Taymans <wi...@fl...>
  *
@@ -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