/* RTP DTMF muxer element for GStreamer * * gstrtpdtmfmux.c: * * Copyright (C) <2007-2010> Nokia Corporation. * Contact: Zeeshan Ali * Copyright (C) <2007-2010> Collabora Ltd * Contact: Olivier Crete * Copyright (C) 1999,2000 Erik Walthinsen * 2000,2005 Wim Taymans * * 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_%%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 #include "config.h" #endif #include #include #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_%d", 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); 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, GstBuffer * buffer); GST_BOILERPLATE (GstRTPDTMFMux, gst_rtp_dtmf_mux, GstRTPMux, GST_TYPE_RTP_MUX); static void gst_rtp_dtmf_mux_init (GstRTPDTMFMux * object, GstRTPDTMFMuxClass * g_class) { } static void gst_rtp_dtmf_mux_base_init (gpointer g_class) { GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&priority_sink_factory)); gst_element_class_set_details_simple (element_class, "RTP muxer", "Codec/Muxer", "mixes RTP DTMF streams into other RTP streams", "Zeeshan Ali "); } static void gst_rtp_dtmf_mux_class_init (GstRTPDTMFMuxClass * klass) { GstElementClass *gstelement_class; GstRTPMuxClass *gstrtpmux_class; gstelement_class = (GstElementClass *) klass; gstrtpmux_class = (GstRTPMuxClass *) klass; 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; } static gboolean gst_rtp_dtmf_mux_accept_buffer_locked (GstRTPMux * rtp_mux, GstRTPMuxPadPrivate * padpriv, GstBuffer * buffer) { GstRTPDTMFMux *mux = GST_RTP_DTMF_MUX (rtp_mux); GstClockTime running_ts; running_ts = GST_BUFFER_TIMESTAMP (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_TIMESTAMP (buffer)); if (padpriv && padpriv->priority) { if (GST_BUFFER_DURATION_IS_VALID (buffer)) { if (GST_CLOCK_TIME_IS_VALID (mux->last_priority_end)) mux->last_priority_end = MAX (running_ts + GST_BUFFER_DURATION (buffer), mux->last_priority_end); else mux->last_priority_end = running_ts + GST_BUFFER_DURATION (buffer); GST_LOG_OBJECT (mux, "Got buffer %p on priority pad, " " blocking regular pads until %" GST_TIME_FORMAT, buffer, GST_TIME_ARGS (mux->last_priority_end)); } else { GST_WARNING_OBJECT (mux, "Buffer %p has an invalid duration," " not blocking other pad", 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, 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", buffer); } return TRUE; } static GstPad * gst_rtp_dtmf_mux_request_new_pad (GstElement * element, GstPadTemplate * templ, const gchar * name) { GstPad *pad; pad = GST_CALL_PARENT_WITH_DEFAULT (GST_ELEMENT_CLASS, request_new_pad, (element, templ, name), NULL); 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_%d") == gst_pad_get_pad_template (pad)) padpriv->priority = TRUE; GST_OBJECT_UNLOCK (element); } return pad; } 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 (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); }