gstreamer/gst/rtsp/gstrtspsrc.c

/* GStreamer
 * Copyright (C) <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.
 */

#ifdef HAVE_CONFIG_H
#include "config.h"
#endif

#include <unistd.h>
#include <string.h>

#include "gstrtspsrc.h"
#include "sdp.h"

/* elementfactory information */
static GstElementDetails gst_rtspsrc_details =
GST_ELEMENT_DETAILS ("RTSP packet receiver",
    "Source/Network",
    "Receive data over the network via RTSP",
    "Wim Taymans <wim@fluendo.com>");

static GstStaticPadTemplate rtptemplate =
GST_STATIC_PAD_TEMPLATE ("rtp_stream%d",
    GST_PAD_SRC,
    GST_PAD_SOMETIMES,
    GST_STATIC_CAPS_ANY);

static GstStaticPadTemplate rtcptemplate =
GST_STATIC_PAD_TEMPLATE ("rtcp_stream%d",
    GST_PAD_SRC,
    GST_PAD_SOMETIMES,
    GST_STATIC_CAPS_ANY);

enum
{
  /* FILL ME */
  LAST_SIGNAL
};

#define DEFAULT_LOCATION	NULL
#define DEFAULT_PROTOCOLS	GST_RTSP_PROTO_UDP_UNICAST | GST_RTSP_PROTO_UDP_MULTICAST | GST_RTSP_PROTO_TCP
#define DEFAULT_DEBUG		FALSE

enum
{
  PROP_0,
  PROP_LOCATION,
  PROP_PROTOCOLS,
  PROP_DEBUG,
  /* FILL ME */
};

#define GST_TYPE_RTSP_PROTO (gst_rtsp_proto_get_type())
static GType
gst_rtsp_proto_get_type (void)
{
  static GType rtsp_proto_type = 0;
  static GFlagsValue rtsp_proto[] = {
    {GST_RTSP_PROTO_UDP_UNICAST, "UDP Unicast", "UDP unicast mode"},
    {GST_RTSP_PROTO_UDP_MULTICAST, "UDP Multicast", "UDP Multicast mode"},
    {GST_RTSP_PROTO_TCP, "TCP", "TCP interleaved mode"},
    {0, NULL, NULL},
  };

  if (!rtsp_proto_type) {
    rtsp_proto_type = g_flags_register_static ("GstRTSPProto", rtsp_proto);
  }
  return rtsp_proto_type;
}


static void gst_rtspsrc_base_init (gpointer g_class);
static void gst_rtspsrc_class_init (GstRTSPSrc * klass);
static void gst_rtspsrc_init (GstRTSPSrc * rtspsrc);

static GstElementStateReturn gst_rtspsrc_change_state (GstElement * element);

static void gst_rtspsrc_set_property (GObject * object, guint prop_id,
    const GValue * value, GParamSpec * pspec);
static void gst_rtspsrc_get_property (GObject * object, guint prop_id,
    GValue * value, GParamSpec * pspec);

static void gst_rtspsrc_loop (GstRTSPSrc * src);

static GstElementClass *parent_class = NULL;

/*static guint gst_rtspsrc_signals[LAST_SIGNAL] = { 0 }; */

GType
gst_rtspsrc_get_type (void)
{
  static GType rtspsrc_type = 0;

  if (!rtspsrc_type) {
    static const GTypeInfo rtspsrc_info = {
      sizeof (GstRTSPSrcClass),
      gst_rtspsrc_base_init,
      NULL,
      (GClassInitFunc) gst_rtspsrc_class_init,
      NULL,
      NULL,
      sizeof (GstRTSPSrc),
      0,
      (GInstanceInitFunc) gst_rtspsrc_init,
      NULL
    };

    rtspsrc_type =
        g_type_register_static (GST_TYPE_ELEMENT, "GstRTSPSrc", &rtspsrc_info,
        0);
  }
  return rtspsrc_type;
}

static void
gst_rtspsrc_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 (&rtptemplate));
  gst_element_class_add_pad_template (element_class,
      gst_static_pad_template_get (&rtcptemplate));

  gst_element_class_set_details (element_class, &gst_rtspsrc_details);
}

static void
gst_rtspsrc_class_init (GstRTSPSrc * klass)
{
  GObjectClass *gobject_class;
  GstElementClass *gstelement_class;

  gobject_class = (GObjectClass *) klass;
  gstelement_class = (GstElementClass *) klass;

  parent_class = g_type_class_ref (GST_TYPE_ELEMENT);

  gobject_class->set_property = gst_rtspsrc_set_property;
  gobject_class->get_property = gst_rtspsrc_get_property;

  g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_LOCATION,
      g_param_spec_string ("location", "RTSP Location",
          "Location of the RTSP url to read",
          DEFAULT_LOCATION, G_PARAM_READWRITE | G_PARAM_CONSTRUCT));

  g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_PROTOCOLS,
      g_param_spec_flags ("protocols", "Protocols", "Allowed protocols",
          GST_TYPE_RTSP_PROTO, DEFAULT_PROTOCOLS,
          G_PARAM_READWRITE | G_PARAM_CONSTRUCT));

  g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_DEBUG,
      g_param_spec_boolean ("debug", "Debug",
          "Dump request and response messages to stdout",
          DEFAULT_DEBUG, G_PARAM_READWRITE | G_PARAM_CONSTRUCT));

  gstelement_class->change_state = gst_rtspsrc_change_state;
}

static void
gst_rtspsrc_init (GstRTSPSrc * src)
{
}

static void
gst_rtspsrc_set_property (GObject * object, guint prop_id, const GValue * value,
    GParamSpec * pspec)
{
  GstRTSPSrc *rtspsrc;

  rtspsrc = GST_RTSPSRC (object);

  switch (prop_id) {
    case PROP_LOCATION:
      g_free (rtspsrc->location);
      rtspsrc->location = g_value_dup_string (value);
      break;
    case PROP_PROTOCOLS:
      rtspsrc->protocols = g_value_get_flags (value);
      break;
    case PROP_DEBUG:
      rtspsrc->debug = g_value_get_boolean (value);
      break;
    default:
      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
      break;
  }
}

static void
gst_rtspsrc_get_property (GObject * object, guint prop_id, GValue * value,
    GParamSpec * pspec)
{
  GstRTSPSrc *rtspsrc;

  rtspsrc = GST_RTSPSRC (object);

  switch (prop_id) {
    case PROP_LOCATION:
      g_value_set_string (value, rtspsrc->location);
      break;
    case PROP_PROTOCOLS:
      g_value_set_flags (value, rtspsrc->protocols);
      break;
    case PROP_DEBUG:
      g_value_set_boolean (value, rtspsrc->debug);
      break;
    default:
      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
      break;
  }
}

static GstRTSPStream *
gst_rtspsrc_create_stream (GstRTSPSrc * src)
{
  GstRTSPStream *s;

  s = g_new0 (GstRTSPStream, 1);
  s->parent = src;
  s->id = src->numstreams++;

  src->streams = g_list_append (src->streams, s);

  return s;
}

static gboolean
gst_rtspsrc_add_element (GstRTSPSrc * src, GstElement * element)
{
  gst_object_set_parent (GST_OBJECT (element), GST_OBJECT (src));
  gst_element_set_manager (element, GST_ELEMENT_MANAGER (src));
  gst_element_set_scheduler (element, GST_ELEMENT_SCHEDULER (src));

  return TRUE;
}

static GstElementStateReturn
gst_rtspsrc_set_state (GstRTSPSrc * src, GstElementState state)
{
  GstElementStateReturn ret;
  GList *streams;

  ret = GST_STATE_SUCCESS;

  /* for all streams */
  for (streams = src->streams; streams; streams = g_list_next (streams)) {
    GstRTSPStream *stream;

    stream = (GstRTSPStream *) streams->data;

    /* first our rtp session manager */
    if ((ret =
            gst_element_set_state (stream->rtpdec, state)) == GST_STATE_FAILURE)
      goto done;

    /* then our sources */
    if (stream->rtpsrc) {
      if ((ret =
              gst_element_set_state (stream->rtpsrc,
                  state)) == GST_STATE_FAILURE)
        goto done;
    }
    if (stream->rtcpsrc) {
      if ((ret =
              gst_element_set_state (stream->rtcpsrc,
                  state)) == GST_STATE_FAILURE)
        goto done;
    }
  }

done:
  return ret;
}

static gboolean
gst_rtspsrc_stream_setup_rtp (GstRTSPStream * stream, gint * rtpport,
    gint * rtcpport)
{
  GstElement *rtpsrc;
  GstElementStateReturn ret;
  GstRTSPSrc *src;

  src = stream->parent;

  if (!(stream->rtpsrc =
          gst_element_make_from_uri (GST_URI_SRC, "udp://0.0.0.0:0", NULL)))
    goto no_udp_rtp_protocol;

  /* we manage this element */
  gst_rtspsrc_add_element (src, stream->rtpsrc);

  ret = gst_element_set_state (stream->rtpsrc, GST_STATE_PAUSED);
  if (ret == GST_STATE_FAILURE)
    goto start_rtp_failure;

  if (!(stream->rtcpsrc =
          gst_element_make_from_uri (GST_URI_SRC, "udp://0.0.0.0:0", NULL)))
    goto no_udp_rtcp_protocol;

  /* we manage this element */
  gst_rtspsrc_add_element (src, stream->rtcpsrc);

  ret = gst_element_set_state (stream->rtcpsrc, GST_STATE_PAUSED);
  if (ret == GST_STATE_FAILURE)
    goto start_rtcp_failure;

  g_object_get (G_OBJECT (stream->rtpsrc), "port", rtpport, NULL);
  g_object_get (G_OBJECT (stream->rtcpsrc), "port", rtcpport, NULL);

  return TRUE;

  /* ERRORS, FIXME, cleanup */
no_udp_rtp_protocol:
  {
    GST_DEBUG ("could not get UDP source for rtp");
    return FALSE;
  }
no_udp_rtcp_protocol:
  {
    GST_DEBUG ("could not get UDP source for rtcp");
    return FALSE;
  }
start_rtp_failure:
  {
    GST_DEBUG ("could not start UDP source for rtp");
    return FALSE;
  }
start_rtcp_failure:
  {
    GST_DEBUG ("could not start UDP source for rtcp");
    return FALSE;
  }
}

static gboolean
gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
    RTSPTransport * transport)
{
  GstRTSPSrc *src;
  GstPad *pad;
  GstElementStateReturn ret;
  gchar *name;

  src = stream->parent;

  if (!(stream->rtpdec = gst_element_factory_make ("rtpdec", NULL)))
    goto no_element;

  /* we manage this element */
  gst_rtspsrc_add_element (src, stream->rtpdec);

  if ((ret =
          gst_element_set_state (stream->rtpdec,
              GST_STATE_PAUSED)) != GST_STATE_SUCCESS)
    goto start_rtpdec_failure;

  stream->rtpdecrtp = gst_element_get_pad (stream->rtpdec, "sinkrtp");
  stream->rtpdecrtcp = gst_element_get_pad (stream->rtpdec, "sinkrtcp");

  /* FIXME, make sure it outputs the caps */
  pad = gst_element_get_pad (stream->rtpdec, "srcrtp");
  name = g_strdup_printf ("rtp_stream%d", stream->id);
  gst_element_add_pad (GST_ELEMENT (src), gst_ghost_pad_new (name, pad));
  g_free (name);
  gst_object_unref (GST_OBJECT (pad));

  if (transport->lower_transport == RTSP_LOWER_TRANS_TCP) {
    /* configure for interleaved delivery, nothing needs to be done
     * here, the loop function will call the chain functions of the
     * rtp session manager. */
  } else {
    /* configure for UDP delivery, we need to connect the udp pads to
     * the rtp session plugin. */
    pad = gst_element_get_pad (stream->rtpsrc, "src");
    gst_pad_link (pad, stream->rtpdecrtp);
    gst_object_unref (GST_OBJECT (pad));

    pad = gst_element_get_pad (stream->rtcpsrc, "src");
    gst_pad_link (pad, stream->rtpdecrtcp);
    gst_object_unref (GST_OBJECT (pad));
  }
  return TRUE;

no_element:
  {
    GST_DEBUG ("no rtpdec element found");
    return FALSE;
  }
start_rtpdec_failure:
  {
    GST_DEBUG ("could not start RTP session");
    return FALSE;
  }
}

static gint
find_stream (GstRTSPStream * stream, gconstpointer a)
{
  gint channel = GPOINTER_TO_INT (a);

  if (stream->rtpchannel == channel || stream->rtcpchannel == channel)
    return 0;

  return -1;
}

static void
gst_rtspsrc_loop (GstRTSPSrc * src)
{
  RTSPMessage response = { 0 };
  RTSPResult res;
  gint channel;
  GList *lstream;
  GstRTSPStream *stream;
  GstPadChainFunction chainfunc;
  GstPad *outpad;
  guint8 *data;
  gint size;

  do {
    GST_DEBUG ("doing reveive");
    if ((res = rtsp_connection_receive (src->connection, &response)) < 0)
      goto receive_error;
    GST_DEBUG ("got packet");
  }
  while (response.type != RTSP_MESSAGE_DATA);

  channel = response.type_data.data.channel;

  lstream = g_list_find_custom (src->streams, GINT_TO_POINTER (channel),
      (GCompareFunc) find_stream);
  if (!lstream)
    goto unknown_stream;

  stream = (GstRTSPStream *) lstream->data;
  if (channel == stream->rtpchannel)
    outpad = stream->rtpdecrtp;
  else if (channel == stream->rtcpchannel)
    outpad = stream->rtpdecrtcp;

  rtsp_message_get_body (&response, &data, &size);

  /* channels are not correct on some servers, do extra check */
  if (data[1] >= 200 && data[1] <= 204) {
    /* hmm RTCP message */
    outpad = stream->rtpdecrtcp;
  }

  /* and chain buffer to internal element */
  {
    GstBuffer *buf;

    buf = gst_buffer_new_and_alloc (size);
    memcpy (GST_BUFFER_DATA (buf), data, size);

    if (gst_pad_chain (outpad, buf) != GST_FLOW_OK)
      goto need_pause;
  }

unknown_stream:

  return;

receive_error:
  {
    GST_ELEMENT_ERROR (src, RESOURCE, WRITE,
        ("Could not receive message."), (NULL));
    /*
       gst_pad_push_event (src->srcpad, gst_event_new (GST_EVENT_EOS));
     */
    goto need_pause;
  }
need_pause:
  {
    gst_task_pause (src->task);
    return;
  }
}

static gboolean
gst_rtspsrc_send (GstRTSPSrc * src, RTSPMessage * request,
    RTSPMessage * response)
{
  RTSPResult res;

  if (src->debug) {
    rtsp_message_dump (request);
  }
  if ((res = rtsp_connection_send (src->connection, request)) < 0)
    goto send_error;

  if ((res = rtsp_connection_receive (src->connection, response)) < 0)
    goto receive_error;
  if (response->type_data.response.code != RTSP_STS_OK)
    goto error_response;

  if (src->debug) {
    rtsp_message_dump (response);
  }

  return TRUE;

send_error:
  {
    GST_ELEMENT_ERROR (src, RESOURCE, WRITE,
        ("Could not send message."), (NULL));
    return FALSE;
  }
receive_error:
  {
    GST_ELEMENT_ERROR (src, RESOURCE, READ,
        ("Could not receive message."), (NULL));
    return FALSE;
  }
error_response:
  {
    rtsp_message_dump (request);
    rtsp_message_dump (response);
    GST_ELEMENT_ERROR (src, RESOURCE, READ, ("Got error response."), (NULL));
    return FALSE;
  }
}

static gboolean
gst_rtspsrc_open (GstRTSPSrc * src)
{
  RTSPUrl *url;
  RTSPResult res;
  RTSPMessage request = { 0 };
  RTSPMessage response = { 0 };
  guint8 *data;
  guint size;
  SDPMessage sdp = { 0 };
  GstRTSPProto protocols;

  /* parse url */
  GST_DEBUG ("parsing url...");
  if ((res = rtsp_url_parse (src->location, &url)) < 0)
    goto invalid_url;

  /* open connection */
  GST_DEBUG ("opening connection...");
  if ((res = rtsp_connection_open (url, &src->connection)) < 0)
    goto could_not_open;

  /* create DESCRIBE */
  GST_DEBUG ("create describe...");
  if ((res =
          rtsp_message_init_request (RTSP_DESCRIBE, src->location,
              &request)) < 0)
    goto create_request_failed;
  /* we accept SDP for now */
  rtsp_message_add_header (&request, RTSP_HDR_ACCEPT, "application/sdp");

  /* send DESCRIBE */
  GST_DEBUG ("send describe...");
  if (!gst_rtspsrc_send (src, &request, &response))
    goto send_error;

  /* parse SDP */
  rtsp_message_get_body (&response, &data, &size);

  GST_DEBUG ("parse sdp...");
  sdp_message_init (&sdp);
  sdp_message_parse_buffer (data, size, &sdp);

  /* we allow all configured protocols */
  protocols = src->protocols;
  /* setup streams */
  {
    gint i;

    for (i = 0; i < sdp_message_medias_len (&sdp); i++) {
      SDPMedia *media;
      gchar *setup_url;
      gchar *control_url;
      gchar *transports;
      GstRTSPStream *stream;

      media = sdp_message_get_media (&sdp, i);

      stream = gst_rtspsrc_create_stream (src);

      GST_DEBUG ("setup media %d", i);
      control_url = sdp_media_get_attribute_val (media, "control");
      if (control_url == NULL) {
        GST_DEBUG ("no control url found, skipping stream");
        continue;
      }

      /* FIXME, check absolute/relative URL */
      setup_url = g_strdup_printf ("%s/%s", src->location, control_url);

      GST_DEBUG ("setup %s", setup_url);
      /* create SETUP request */
      if ((res =
              rtsp_message_init_request (RTSP_SETUP, setup_url,
                  &request)) < 0) {
        g_free (setup_url);
        goto create_request_failed;
      }
      g_free (setup_url);


      transports = g_strdup ("");
      if (protocols & GST_RTSP_PROTO_UDP_UNICAST) {
        gchar *new;
        gint rtpport, rtcpport;
        gchar *trxparams;
        gboolean res;

        /* allocate two udp ports */
        if (!gst_rtspsrc_stream_setup_rtp (stream, &rtpport, &rtcpport))
          goto setup_rtp_failed;

        trxparams = g_strdup_printf ("client_port=%d-%d", rtpport, rtcpport);
        new = g_strconcat (transports, "RTP/AVP/UDP;unicast;", trxparams, NULL);
        g_free (trxparams);
        g_free (transports);
        transports = new;
      }
      if (protocols & GST_RTSP_PROTO_UDP_MULTICAST) {
        gchar *new;

        new =
            g_strconcat (transports, transports[0] ? "," : "",
            "RTP/AVP/UDP;multicast", NULL);
        g_free (transports);
        transports = new;
      }
      if (protocols & GST_RTSP_PROTO_TCP) {
        gchar *new;

        new =
            g_strconcat (transports, transports[0] ? "," : "", "RTP/AVP/TCP",
            NULL);
        g_free (transports);
        transports = new;
      }

      /* select transport, copy is made when adding to header */
      rtsp_message_add_header (&request, RTSP_HDR_TRANSPORT, transports);
      g_free (transports);

      if (!gst_rtspsrc_send (src, &request, &response))
        goto send_error;

      /* parse response transport */
      {
        gchar *resptrans;
        RTSPTransport transport = { 0 };

        rtsp_message_get_header (&response, RTSP_HDR_TRANSPORT, &resptrans);

        /* parse transport */
        rtsp_transport_parse (resptrans, &transport);
        /* update allowed transports for other streams */
        if (transport.lower_transport == RTSP_LOWER_TRANS_TCP) {
          protocols = GST_RTSP_PROTO_TCP;
          src->interleaved = TRUE;
        } else {
          if (transport.multicast) {
            /* disable unicast */
            protocols = GST_RTSP_PROTO_UDP_MULTICAST;
          } else {
            /* disable multicast */
            protocols = GST_RTSP_PROTO_UDP_UNICAST;
          }
        }
        /* now configure the stream with the transport */
        if (!gst_rtspsrc_stream_configure_transport (stream, &transport)) {
          GST_DEBUG ("could not configure stream transport, skipping stream");
        }
        /* clean up our transport struct */
        rtsp_transport_init (&transport);
      }
    }
  }
  return TRUE;

  /* ERRORS */
invalid_url:
  {
    GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND,
        ("Not a valid RTSP url."), (NULL));
    return FALSE;
  }
could_not_open:
  {
    GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE,
        ("Could not open connection."), (NULL));
    return FALSE;
  }
create_request_failed:
  {
    GST_ELEMENT_ERROR (src, LIBRARY, INIT,
        ("Could not create request."), (NULL));
    return FALSE;
  }
send_error:
  {
    GST_ELEMENT_ERROR (src, RESOURCE, WRITE,
        ("Could not send message."), (NULL));
    return FALSE;
  }
setup_rtp_failed:
  {
    GST_ELEMENT_ERROR (src, RESOURCE, WRITE, ("Could not setup rtp."), (NULL));
    return FALSE;
  }
}

static gboolean
gst_rtspsrc_close (GstRTSPSrc * src)
{
  RTSPMessage request = { 0 };
  RTSPMessage response = { 0 };
  RTSPResult res;

  GST_DEBUG ("TEARDOWN...");

  /* stop task if any */
  if (src->task) {
    gst_task_stop (src->task);
    gst_object_unref (GST_OBJECT (src->task));
    src->task = NULL;
  }

  /* do TEARDOWN */
  if ((res =
          rtsp_message_init_request (RTSP_TEARDOWN, src->location,
              &request)) < 0)
    goto create_request_failed;

  if (!gst_rtspsrc_send (src, &request, &response))
    goto send_error;

  /* close connection */
  GST_DEBUG ("closing connection...");
  if ((res = rtsp_connection_close (src->connection)) < 0)
    goto close_failed;

  return TRUE;

create_request_failed:
  {
    GST_ELEMENT_ERROR (src, LIBRARY, INIT,
        ("Could not create request."), (NULL));
    return FALSE;
  }
send_error:
  {
    GST_ELEMENT_ERROR (src, RESOURCE, WRITE,
        ("Could not send message."), (NULL));
    return FALSE;
  }
close_failed:
  {
    GST_ELEMENT_ERROR (src, RESOURCE, CLOSE, ("Close failed."), (NULL));
    return FALSE;
  }
}

static gboolean
gst_rtspsrc_play (GstRTSPSrc * src)
{
  RTSPMessage request = { 0 };
  RTSPMessage response = { 0 };
  RTSPResult res;

  GST_DEBUG ("PLAY...");

  /* do play */
  if ((res =
          rtsp_message_init_request (RTSP_PLAY, src->location, &request)) < 0)
    goto create_request_failed;

  if (!gst_rtspsrc_send (src, &request, &response))
    goto send_error;

  if (GST_ELEMENT_SCHEDULER (src) && src->interleaved) {
    src->task =
        gst_scheduler_create_task (GST_ELEMENT_SCHEDULER (src),
        (GstTaskFunction) gst_rtspsrc_loop, src);

    gst_task_start (src->task);
  }

  return TRUE;

create_request_failed:
  {
    GST_ELEMENT_ERROR (src, LIBRARY, INIT,
        ("Could not create request."), (NULL));
    return FALSE;
  }
send_error:
  {
    GST_ELEMENT_ERROR (src, RESOURCE, WRITE,
        ("Could not send message."), (NULL));
    return FALSE;
  }
}

static gboolean
gst_rtspsrc_pause (GstRTSPSrc * src)
{
  RTSPMessage request = { 0 };
  RTSPMessage response = { 0 };
  RTSPResult res;

  GST_DEBUG ("PAUSE...");
  /* do pause */
  if ((res =
          rtsp_message_init_request (RTSP_PAUSE, src->location, &request)) < 0)
    goto create_request_failed;

  if (!gst_rtspsrc_send (src, &request, &response))
    goto send_error;

  return TRUE;

create_request_failed:
  {
    GST_ELEMENT_ERROR (src, LIBRARY, INIT,
        ("Could not create request."), (NULL));
    return FALSE;
  }
send_error:
  {
    GST_ELEMENT_ERROR (src, RESOURCE, WRITE,
        ("Could not send message."), (NULL));
    return FALSE;
  }
}

static GstElementStateReturn
gst_rtspsrc_change_state (GstElement * element)
{
  GstRTSPSrc *rtspsrc;
  GstElementState transition;
  GstElementStateReturn ret;

  rtspsrc = GST_RTSPSRC (element);

  transition = GST_STATE_TRANSITION (rtspsrc);

  switch (transition) {
    case GST_STATE_NULL_TO_READY:
      break;
    case GST_STATE_READY_TO_PAUSED:
      rtspsrc->interleaved = FALSE;
      if (!gst_rtspsrc_open (rtspsrc))
        goto open_failed;
      break;
    case GST_STATE_PAUSED_TO_PLAYING:
      gst_rtspsrc_play (rtspsrc);
      break;
    default:
      break;
  }

  ret = GST_ELEMENT_CLASS (parent_class)->change_state (element);
  if (ret == GST_STATE_FAILURE)
    goto done;

  ret = gst_rtspsrc_set_state (rtspsrc, GST_STATE_PENDING (rtspsrc));
  if (ret == GST_STATE_FAILURE)
    goto done;

  switch (transition) {
    case GST_STATE_PLAYING_TO_PAUSED:
      gst_rtspsrc_pause (rtspsrc);
      break;
    case GST_STATE_PAUSED_TO_READY:
      break;
    case GST_STATE_READY_TO_NULL:
      break;
    default:
      break;
  }

done:
  return ret;

open_failed:
  {
    return GST_STATE_FAILURE;
  }
}