diff options
| author | Olivier Crête <olivier.crete@collabora.co.uk> | 2010-09-30 16:07:29 -0400 |
|---|---|---|
| committer | Olivier Crête <olivier.crete@collabora.co.uk> | 2010-09-30 16:07:29 -0400 |
| commit | 51887b934f701497a8842959447c0d8d1610f9e9 (patch) | |
| tree | c0ccd660f8447380efa4569ad550912b1813ff8f /gst/rtpmux | |
| parent | 5cb8164938cc4b902ee697a5754e203464fed9e1 (diff) | |
| download | gstreamer-plugins-bad-51887b934f701497a8842959447c0d8d1610f9e9.tar.gz | |
rtpmux: Improve documentation
Add an example pipeline, and try to explain a bit more what it does.
Diffstat (limited to 'gst/rtpmux')
| -rw-r--r-- | gst/rtpmux/gstrtpdtmfmux.c | 4 | ||||
| -rw-r--r-- | 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 c95391011..f62c6263d 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 239b6d306..f01440093 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 <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 |