mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-04 23:46:43 +00:00
234 lines
7.5 KiB
C
234 lines
7.5 KiB
C
/* 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., 51 Franklin St, Fifth Floor,
|
|
* Boston, MA 02110-1301, USA.
|
|
*/
|
|
|
|
/**
|
|
* SECTION:element-rtpdtmfmux
|
|
* @title: rtpdtmfmux
|
|
* @see_also: rtpdtmfsrc, dtmfsrc, rtpmux
|
|
*
|
|
* The RTP "DTMF" Muxer muxes multiple RTP streams into a valid RTP
|
|
* stream. It does exactly what its 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_static_pad_template (gstelement_class,
|
|
&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_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);
|
|
}
|