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/* RTP DTMF muxer element for GStreamer
*
* gstrtpdtmfmux.c:
*
* 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>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/**
* SECTION:element-rtpdtmfmux
* @see_also: rtpdtmfsrc, dtmfsrc, rtpmux
*
* The RTP "DTMF" Muxer muxes multiple RTP streams into a valid RTP
* stream. It does exactly what it's parent (#rtpmux) does, except
* that it prevent buffers coming over a regular sink_%%u pad from going through
* for the duration of buffers that came in a priority_sink_%%u 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
#include "config.h"
#endif
#include <gst/gst.h>
#include <string.h>
#include "gstrtpdtmfmux.h"
GST_DEBUG_CATEGORY_STATIC (gst_rtp_dtmf_mux_debug);
#define GST_CAT_DEFAULT gst_rtp_dtmf_mux_debug
static GstStaticPadTemplate priority_sink_factory =
GST_STATIC_PAD_TEMPLATE ("priority_sink_%u",
GST_PAD_SINK,
GST_PAD_REQUEST,
GST_STATIC_CAPS ("application/x-rtp"));
static GstPad *gst_rtp_dtmf_mux_request_new_pad (GstElement * element,
GstPadTemplate * templ, const gchar * name, const GstCaps * caps);
static GstStateChangeReturn gst_rtp_dtmf_mux_change_state (GstElement * element,
GstStateChange transition);
static gboolean gst_rtp_dtmf_mux_accept_buffer_locked (GstRTPMux * rtp_mux,
GstRTPMuxPadPrivate * padpriv, GstRTPBuffer * rtpbuffer);
static gboolean gst_rtp_dtmf_mux_src_event (GstRTPMux * rtp_mux,
GstEvent * event);
G_DEFINE_TYPE (GstRTPDTMFMux, gst_rtp_dtmf_mux, GST_TYPE_RTP_MUX);
static void
gst_rtp_dtmf_mux_init (GstRTPDTMFMux * mux)
{
}
static void
gst_rtp_dtmf_mux_class_init (GstRTPDTMFMuxClass * klass)
{
GstElementClass *gstelement_class;
GstRTPMuxClass *gstrtpmux_class;
gstelement_class = (GstElementClass *) klass;
gstrtpmux_class = (GstRTPMuxClass *) klass;
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&priority_sink_factory));
gst_element_class_set_static_metadata (gstelement_class, "RTP muxer",
"Codec/Muxer",
"mixes RTP DTMF streams into other RTP streams",
"Zeeshan Ali <first.last@nokia.com>");
gstelement_class->request_new_pad =
GST_DEBUG_FUNCPTR (gst_rtp_dtmf_mux_request_new_pad);
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_rtp_dtmf_mux_change_state);
gstrtpmux_class->accept_buffer_locked = gst_rtp_dtmf_mux_accept_buffer_locked;
gstrtpmux_class->src_event = gst_rtp_dtmf_mux_src_event;
}
static gboolean
gst_rtp_dtmf_mux_accept_buffer_locked (GstRTPMux * rtp_mux,
GstRTPMuxPadPrivate * padpriv, GstRTPBuffer * rtpbuffer)
{
GstRTPDTMFMux *mux = GST_RTP_DTMF_MUX (rtp_mux);
GstClockTime running_ts;
running_ts = GST_BUFFER_PTS (rtpbuffer->buffer);
if (GST_CLOCK_TIME_IS_VALID (running_ts)) {
if (padpriv && padpriv->segment.format == GST_FORMAT_TIME)
running_ts = gst_segment_to_running_time (&padpriv->segment,
GST_FORMAT_TIME, GST_BUFFER_PTS (rtpbuffer->buffer));
if (padpriv && padpriv->priority) {
if (GST_BUFFER_PTS_IS_VALID (rtpbuffer->buffer)) {
if (GST_CLOCK_TIME_IS_VALID (mux->last_priority_end))
mux->last_priority_end =
MAX (running_ts + GST_BUFFER_DURATION (rtpbuffer->buffer),
mux->last_priority_end);
else
mux->last_priority_end = running_ts +
GST_BUFFER_DURATION (rtpbuffer->buffer);
GST_LOG_OBJECT (mux, "Got buffer %p on priority pad, "
" blocking regular pads until %" GST_TIME_FORMAT, rtpbuffer->buffer,
GST_TIME_ARGS (mux->last_priority_end));
} else {
GST_WARNING_OBJECT (mux, "Buffer %p has an invalid duration,"
" not blocking other pad", rtpbuffer->buffer);
}
} else {
if (GST_CLOCK_TIME_IS_VALID (mux->last_priority_end) &&
running_ts < mux->last_priority_end) {
GST_LOG_OBJECT (mux, "Dropping buffer %p because running time"
" %" GST_TIME_FORMAT " < %" GST_TIME_FORMAT, rtpbuffer->buffer,
GST_TIME_ARGS (running_ts), GST_TIME_ARGS (mux->last_priority_end));
return FALSE;
}
}
} else {
GST_LOG_OBJECT (mux, "Buffer %p has an invalid timestamp,"
" letting through", rtpbuffer->buffer);
}
return TRUE;
}
static GstPad *
gst_rtp_dtmf_mux_request_new_pad (GstElement * element, GstPadTemplate * templ,
const gchar * name, const GstCaps * caps)
{
GstPad *pad;
pad =
GST_ELEMENT_CLASS (gst_rtp_dtmf_mux_parent_class)->request_new_pad
(element, templ, name, caps);
if (pad) {
GstRTPMuxPadPrivate *padpriv;
GST_OBJECT_LOCK (element);
padpriv = gst_pad_get_element_private (pad);
if (gst_element_class_get_pad_template (GST_ELEMENT_GET_CLASS (element),
"priority_sink_%u") == gst_pad_get_pad_template (pad))
padpriv->priority = TRUE;
GST_OBJECT_UNLOCK (element);
}
return pad;
}
static gboolean
gst_rtp_dtmf_mux_src_event (GstRTPMux * rtp_mux, GstEvent * event)
{
if (GST_EVENT_TYPE (event) == GST_EVENT_CUSTOM_UPSTREAM) {
const GstStructure *s = gst_event_get_structure (event);
if (s && gst_structure_has_name (s, "dtmf-event")) {
GST_OBJECT_LOCK (rtp_mux);
if (GST_CLOCK_TIME_IS_VALID (rtp_mux->last_stop)) {
event = (GstEvent *)
gst_mini_object_make_writable (GST_MINI_OBJECT_CAST (event));
s = gst_event_get_structure (event);
gst_structure_set ((GstStructure *) s,
"last-stop", G_TYPE_UINT64, rtp_mux->last_stop, NULL);
}
GST_OBJECT_UNLOCK (rtp_mux);
}
}
return GST_RTP_MUX_CLASS (gst_rtp_dtmf_mux_parent_class)->src_event (rtp_mux,
event);
}
static GstStateChangeReturn
gst_rtp_dtmf_mux_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret;
GstRTPDTMFMux *mux = GST_RTP_DTMF_MUX (element);
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
{
GST_OBJECT_LOCK (mux);
mux->last_priority_end = GST_CLOCK_TIME_NONE;
GST_OBJECT_UNLOCK (mux);
break;
}
default:
break;
}
ret =
GST_ELEMENT_CLASS (gst_rtp_dtmf_mux_parent_class)->change_state (element,
transition);
return ret;
}
gboolean
gst_rtp_dtmf_mux_plugin_init (GstPlugin * plugin)
{
GST_DEBUG_CATEGORY_INIT (gst_rtp_dtmf_mux_debug, "rtpdtmfmux", 0,
"rtp dtmf muxer");
return gst_element_register (plugin, "rtpdtmfmux", GST_RANK_NONE,
GST_TYPE_RTP_DTMF_MUX);
}
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