/* 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

 * @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);

}