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234 lines
7.5 KiB
C
234 lines
7.5 KiB
C
/* RTP DTMF muxer element for GStreamer
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*
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* gstrtpdtmfmux.c:
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*
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* Copyright (C) <2007-2010> Nokia Corporation.
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* Contact: Zeeshan Ali <zeeshan.ali@nokia.com>
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* Copyright (C) <2007-2010> Collabora Ltd
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* Contact: Olivier Crete <olivier.crete@collabora.co.uk>
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* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
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* 2000,2005 Wim Taymans <wim@fluendo.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:element-rtpdtmfmux
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* @title: rtpdtmfmux
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* @see_also: rtpdtmfsrc, dtmfsrc, rtpmux
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*
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* The RTP "DTMF" Muxer muxes multiple RTP streams into a valid RTP
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* stream. It does exactly what its parent (#rtpmux) does, except
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* that it prevent buffers coming over a regular sink_\%u pad from going through
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* for the duration of buffers that came in a priority_sink_\%u pad.
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*
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* This is especially useful if a discontinuous source like dtmfsrc or
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* rtpdtmfsrc are connected to the priority sink pads. This way, the generated
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* DTMF signal can replace the recorded audio while the tone is being sent.
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <gst/gst.h>
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#include <string.h>
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#include "gstrtpdtmfmux.h"
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GST_DEBUG_CATEGORY_STATIC (gst_rtp_dtmf_mux_debug);
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#define GST_CAT_DEFAULT gst_rtp_dtmf_mux_debug
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static GstStaticPadTemplate priority_sink_factory =
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GST_STATIC_PAD_TEMPLATE ("priority_sink_%u",
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GST_PAD_SINK,
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GST_PAD_REQUEST,
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GST_STATIC_CAPS ("application/x-rtp"));
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static GstPad *gst_rtp_dtmf_mux_request_new_pad (GstElement * element,
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GstPadTemplate * templ, const gchar * name, const GstCaps * caps);
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static GstStateChangeReturn gst_rtp_dtmf_mux_change_state (GstElement * element,
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GstStateChange transition);
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static gboolean gst_rtp_dtmf_mux_accept_buffer_locked (GstRTPMux * rtp_mux,
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GstRTPMuxPadPrivate * padpriv, GstRTPBuffer * rtpbuffer);
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static gboolean gst_rtp_dtmf_mux_src_event (GstRTPMux * rtp_mux,
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GstEvent * event);
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G_DEFINE_TYPE (GstRTPDTMFMux, gst_rtp_dtmf_mux, GST_TYPE_RTP_MUX);
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static void
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gst_rtp_dtmf_mux_init (GstRTPDTMFMux * mux)
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{
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}
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static void
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gst_rtp_dtmf_mux_class_init (GstRTPDTMFMuxClass * klass)
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{
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GstElementClass *gstelement_class;
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GstRTPMuxClass *gstrtpmux_class;
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gstelement_class = (GstElementClass *) klass;
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gstrtpmux_class = (GstRTPMuxClass *) klass;
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gst_element_class_add_static_pad_template (gstelement_class,
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&priority_sink_factory);
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gst_element_class_set_static_metadata (gstelement_class, "RTP muxer",
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"Codec/Muxer",
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"mixes RTP DTMF streams into other RTP streams",
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"Zeeshan Ali <first.last@nokia.com>");
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gstelement_class->request_new_pad =
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GST_DEBUG_FUNCPTR (gst_rtp_dtmf_mux_request_new_pad);
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gstelement_class->change_state =
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GST_DEBUG_FUNCPTR (gst_rtp_dtmf_mux_change_state);
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gstrtpmux_class->accept_buffer_locked = gst_rtp_dtmf_mux_accept_buffer_locked;
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gstrtpmux_class->src_event = gst_rtp_dtmf_mux_src_event;
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}
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static gboolean
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gst_rtp_dtmf_mux_accept_buffer_locked (GstRTPMux * rtp_mux,
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GstRTPMuxPadPrivate * padpriv, GstRTPBuffer * rtpbuffer)
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{
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GstRTPDTMFMux *mux = GST_RTP_DTMF_MUX (rtp_mux);
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GstClockTime running_ts;
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running_ts = GST_BUFFER_PTS (rtpbuffer->buffer);
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if (GST_CLOCK_TIME_IS_VALID (running_ts)) {
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if (padpriv && padpriv->segment.format == GST_FORMAT_TIME)
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running_ts = gst_segment_to_running_time (&padpriv->segment,
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GST_FORMAT_TIME, GST_BUFFER_PTS (rtpbuffer->buffer));
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if (padpriv && padpriv->priority) {
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if (GST_BUFFER_PTS_IS_VALID (rtpbuffer->buffer)) {
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if (GST_CLOCK_TIME_IS_VALID (mux->last_priority_end))
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mux->last_priority_end =
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MAX (running_ts + GST_BUFFER_DURATION (rtpbuffer->buffer),
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mux->last_priority_end);
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else
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mux->last_priority_end = running_ts +
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GST_BUFFER_DURATION (rtpbuffer->buffer);
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GST_LOG_OBJECT (mux, "Got buffer %p on priority pad, "
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" blocking regular pads until %" GST_TIME_FORMAT, rtpbuffer->buffer,
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GST_TIME_ARGS (mux->last_priority_end));
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} else {
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GST_WARNING_OBJECT (mux, "Buffer %p has an invalid duration,"
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" not blocking other pad", rtpbuffer->buffer);
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}
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} else {
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if (GST_CLOCK_TIME_IS_VALID (mux->last_priority_end) &&
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running_ts < mux->last_priority_end) {
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GST_LOG_OBJECT (mux, "Dropping buffer %p because running time"
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" %" GST_TIME_FORMAT " < %" GST_TIME_FORMAT, rtpbuffer->buffer,
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GST_TIME_ARGS (running_ts), GST_TIME_ARGS (mux->last_priority_end));
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return FALSE;
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}
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}
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} else {
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GST_LOG_OBJECT (mux, "Buffer %p has an invalid timestamp,"
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" letting through", rtpbuffer->buffer);
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}
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return TRUE;
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}
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static GstPad *
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gst_rtp_dtmf_mux_request_new_pad (GstElement * element, GstPadTemplate * templ,
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const gchar * name, const GstCaps * caps)
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{
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GstPad *pad;
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pad =
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GST_ELEMENT_CLASS (gst_rtp_dtmf_mux_parent_class)->request_new_pad
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(element, templ, name, caps);
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if (pad) {
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GstRTPMuxPadPrivate *padpriv;
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GST_OBJECT_LOCK (element);
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padpriv = gst_pad_get_element_private (pad);
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if (gst_element_class_get_pad_template (GST_ELEMENT_GET_CLASS (element),
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"priority_sink_%u") == GST_PAD_PAD_TEMPLATE (pad))
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padpriv->priority = TRUE;
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GST_OBJECT_UNLOCK (element);
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}
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return pad;
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}
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static gboolean
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gst_rtp_dtmf_mux_src_event (GstRTPMux * rtp_mux, GstEvent * event)
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{
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if (GST_EVENT_TYPE (event) == GST_EVENT_CUSTOM_UPSTREAM) {
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const GstStructure *s = gst_event_get_structure (event);
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if (s && gst_structure_has_name (s, "dtmf-event")) {
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GST_OBJECT_LOCK (rtp_mux);
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if (GST_CLOCK_TIME_IS_VALID (rtp_mux->last_stop)) {
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event = (GstEvent *)
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gst_mini_object_make_writable (GST_MINI_OBJECT_CAST (event));
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s = gst_event_get_structure (event);
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gst_structure_set ((GstStructure *) s,
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"last-stop", G_TYPE_UINT64, rtp_mux->last_stop, NULL);
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}
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GST_OBJECT_UNLOCK (rtp_mux);
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}
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}
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return GST_RTP_MUX_CLASS (gst_rtp_dtmf_mux_parent_class)->src_event (rtp_mux,
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event);
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}
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static GstStateChangeReturn
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gst_rtp_dtmf_mux_change_state (GstElement * element, GstStateChange transition)
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{
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GstStateChangeReturn ret;
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GstRTPDTMFMux *mux = GST_RTP_DTMF_MUX (element);
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switch (transition) {
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case GST_STATE_CHANGE_READY_TO_PAUSED:
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{
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GST_OBJECT_LOCK (mux);
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mux->last_priority_end = GST_CLOCK_TIME_NONE;
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GST_OBJECT_UNLOCK (mux);
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break;
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}
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default:
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break;
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}
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ret =
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GST_ELEMENT_CLASS (gst_rtp_dtmf_mux_parent_class)->change_state (element,
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transition);
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return ret;
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}
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gboolean
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gst_rtp_dtmf_mux_plugin_init (GstPlugin * plugin)
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{
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GST_DEBUG_CATEGORY_INIT (gst_rtp_dtmf_mux_debug, "rtpdtmfmux", 0,
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"rtp dtmf muxer");
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return gst_element_register (plugin, "rtpdtmfmux", GST_RANK_NONE,
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GST_TYPE_RTP_DTMF_MUX);
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}
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