gstreamer/gst/rtsp-server/rtsp-media.c

/* GStreamer
 * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.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.
 */

#include "rtsp-media.h"

static void gst_rtsp_media_finalize (GObject * obj);

G_DEFINE_TYPE (GstRTSPMedia, gst_rtsp_media, G_TYPE_OBJECT);

static void
gst_rtsp_media_class_init (GstRTSPMediaClass * klass)
{
  GObjectClass *gobject_class;

  gobject_class = G_OBJECT_CLASS (klass);

  gobject_class->finalize = gst_rtsp_media_finalize;
}

static void
gst_rtsp_media_init (GstRTSPMedia * media)
{
  media->streams = g_array_new (FALSE, TRUE, sizeof (GstRTSPMediaStream *));
}

static void
gst_rtsp_media_stream_free (GstRTSPMediaStream *stream)
{
}

static void
gst_rtsp_media_finalize (GObject * obj)
{
  GstRTSPMedia *media;
  guint i;

  media = GST_RTSP_MEDIA (obj);

  for (i = 0; i < media->streams->len; i++) {
    GstRTSPMediaStream *stream;

    stream = g_array_index (media->streams, GstRTSPMediaStream *, i);

    gst_rtsp_media_stream_free (stream);
  }
  g_array_free (media->streams, TRUE);

  G_OBJECT_CLASS (gst_rtsp_media_parent_class)->finalize (obj);
}

/**
 * gst_rtsp_media_new:
 *
 * Create a new #GstRTSPMedia instance. The #GstRTSPMedia object contains the
 * element to produde RTP data for one or more related (audio/video/..) 
 * streams.
 *
 * Returns: a new #GstRTSPMedia object.
 */
GstRTSPMedia *
gst_rtsp_media_new (void)
{
  GstRTSPMedia *result;

  result = g_object_new (GST_TYPE_RTSP_MEDIA, NULL);

  return result;
}


/**
 * gst_rtsp_media_n_streams:
 * @media: a #GstRTSPMedia
 *
 * Get the number of streams in this media.
 *
 * Returns: The number of streams.
 */
guint
gst_rtsp_media_n_streams (GstRTSPMedia *media)
{
  g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), 0);

  return media->streams->len;
}

/**
 * gst_rtsp_media_get_stream:
 * @media: a #GstRTSPMedia
 * @idx: the stream index
 *
 * Retrieve the stream with index @idx from @media.
 *
 * Returns: the #GstRTSPMediaStream at index @idx.
 */
GstRTSPMediaStream *
gst_rtsp_media_get_stream (GstRTSPMedia *media, guint idx)
{
  GstRTSPMediaStream *res;
  
  g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
  g_return_val_if_fail (idx < media->streams->len, NULL);

  res = g_array_index (media->streams, GstRTSPMediaStream *, idx);

  return res;
}

/* Allocate the udp ports and sockets */
static gboolean
alloc_udp_ports (GstRTSPMediaStream * stream)
{
  GstStateChangeReturn ret;
  GstElement *udpsrc0, *udpsrc1;
  GstElement *udpsink0, *udpsink1;
  gint tmp_rtp, tmp_rtcp;
  guint count;
  gint rtpport, rtcpport, sockfd;

  udpsrc0 = NULL;
  udpsrc1 = NULL;
  udpsink0 = NULL;
  udpsink1 = NULL;
  count = 0;

  /* Start with random port */
  tmp_rtp = 0;

  /* try to allocate 2 UDP ports, the RTP port should be an even
   * number and the RTCP port should be the next (uneven) port */
again:
  udpsrc0 = gst_element_make_from_uri (GST_URI_SRC, "udp://0.0.0.0", NULL);
  if (udpsrc0 == NULL)
    goto no_udp_protocol;
  g_object_set (G_OBJECT (udpsrc0), "port", tmp_rtp, NULL);

  ret = gst_element_set_state (udpsrc0, GST_STATE_PAUSED);
  if (ret == GST_STATE_CHANGE_FAILURE) {
    if (tmp_rtp != 0) {
      tmp_rtp += 2;
      if (++count > 20)
        goto no_ports;

      gst_element_set_state (udpsrc0, GST_STATE_NULL);
      gst_object_unref (udpsrc0);

      goto again;
    }
    goto no_udp_protocol;
  }

  g_object_get (G_OBJECT (udpsrc0), "port", &tmp_rtp, NULL);

  /* check if port is even */
  if ((tmp_rtp & 1) != 0) {
    /* port not even, close and allocate another */
    if (++count > 20)
      goto no_ports;

    gst_element_set_state (udpsrc0, GST_STATE_NULL);
    gst_object_unref (udpsrc0);

    tmp_rtp++;
    goto again;
  }

  /* allocate port+1 for RTCP now */
  udpsrc1 = gst_element_make_from_uri (GST_URI_SRC, "udp://0.0.0.0", NULL);
  if (udpsrc1 == NULL)
    goto no_udp_rtcp_protocol;

  /* set port */
  tmp_rtcp = tmp_rtp + 1;
  g_object_set (G_OBJECT (udpsrc1), "port", tmp_rtcp, NULL);

  ret = gst_element_set_state (udpsrc1, GST_STATE_PAUSED);
  /* tmp_rtcp port is busy already : retry to make rtp/rtcp pair */
  if (ret == GST_STATE_CHANGE_FAILURE) {

    if (++count > 20)
      goto no_ports;

    gst_element_set_state (udpsrc0, GST_STATE_NULL);
    gst_object_unref (udpsrc0);

    gst_element_set_state (udpsrc1, GST_STATE_NULL);
    gst_object_unref (udpsrc1);

    tmp_rtp += 2;
    goto again;
  }

  /* all fine, do port check */
  g_object_get (G_OBJECT (udpsrc0), "port", &rtpport, NULL);
  g_object_get (G_OBJECT (udpsrc1), "port", &rtcpport, NULL);

  /* this should not happen... */
  if (rtpport != tmp_rtp || rtcpport != tmp_rtcp)
    goto port_error;

  udpsink0 = gst_element_factory_make ("multiudpsink", NULL);
  if (!udpsink0)
    goto no_udp_protocol;

  g_object_get (G_OBJECT (udpsrc0), "sock", &sockfd, NULL);
  g_object_set (G_OBJECT (udpsink0), "sockfd", sockfd, NULL);
  g_object_set (G_OBJECT (udpsink0), "closefd", FALSE, NULL);

  udpsink1 = gst_element_factory_make ("multiudpsink", NULL);
  if (!udpsink1)
    goto no_udp_protocol;

  g_object_get (G_OBJECT (udpsrc1), "sock", &sockfd, NULL);
  g_object_set (G_OBJECT (udpsink1), "sockfd", sockfd, NULL);
  g_object_set (G_OBJECT (udpsink1), "closefd", FALSE, NULL);
  g_object_set (G_OBJECT (udpsink1), "sync", FALSE, NULL);
  g_object_set (G_OBJECT (udpsink1), "async", FALSE, NULL);

  /* we keep these elements, we configure all in configure_transport when the
   * server told us to really use the UDP ports. */
  stream->udpsrc[0] = gst_object_ref (udpsrc0);
  stream->udpsrc[1] = gst_object_ref (udpsrc1);
  stream->udpsink[0] = gst_object_ref (udpsink0);
  stream->udpsink[1] = gst_object_ref (udpsink1);
  stream->server_port.min = rtpport;
  stream->server_port.max = rtcpport;

  /* they are ours now */
  gst_object_sink (udpsrc0);
  gst_object_sink (udpsrc1);
  gst_object_sink (udpsink0);
  gst_object_sink (udpsink1);

  return TRUE;

  /* ERRORS */
no_udp_protocol:
  {
    goto cleanup;
  }
no_ports:
  {
    goto cleanup;
  }
no_udp_rtcp_protocol:
  {
    goto cleanup;
  }
port_error:
  {
    goto cleanup;
  }
cleanup:
  {
    if (udpsrc0) {
      gst_element_set_state (udpsrc0, GST_STATE_NULL);
      gst_object_unref (udpsrc0);
    }
    if (udpsrc1) {
      gst_element_set_state (udpsrc1, GST_STATE_NULL);
      gst_object_unref (udpsrc1);
    }
    if (udpsink0) {
      gst_element_set_state (udpsink0, GST_STATE_NULL);
      gst_object_unref (udpsink0);
    }
    if (udpsink1) {
      gst_element_set_state (udpsink1, GST_STATE_NULL);
      gst_object_unref (udpsink1);
    }
    return FALSE;
  }
}

static void
caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPMediaStream * stream)
{
  gchar *capsstr;

  if (stream->caps)
    gst_caps_unref (stream->caps);
  if ((stream->caps = GST_PAD_CAPS (pad)))
    gst_caps_ref (stream->caps);

  capsstr = gst_caps_to_string (stream->caps);
  g_message ("stream %p received caps %s", stream, capsstr);
  g_free (capsstr);
}

/* prepare the pipeline objects to handle @stream in @media */
static gboolean
setup_stream (GstRTSPMediaStream *stream, GstRTSPMedia *media)
{
  gchar *name;
  GstPad *pad;

  alloc_udp_ports (stream);

  gst_bin_add (GST_BIN_CAST (media->pipeline), stream->udpsink[0]);
  gst_bin_add (GST_BIN_CAST (media->pipeline), stream->udpsink[1]);
  gst_bin_add (GST_BIN_CAST (media->pipeline), stream->udpsrc[1]);

  /* hook up the stream to the RTP session elements. */
  name = g_strdup_printf ("send_rtp_sink_%d", stream->idx);
  stream->send_rtp_sink = gst_element_get_request_pad (media->rtpbin, name);
  g_free (name);
  name = g_strdup_printf ("send_rtp_src_%d", stream->idx);
  stream->send_rtp_src = gst_element_get_static_pad (media->rtpbin, name);
  g_free (name);
  name = g_strdup_printf ("send_rtcp_src_%d", stream->idx);
  stream->send_rtcp_src = gst_element_get_request_pad (media->rtpbin, name);
  g_free (name);
  name = g_strdup_printf ("recv_rtcp_sink_%d", stream->idx);
  stream->recv_rtcp_sink = gst_element_get_request_pad (media->rtpbin, name);
  g_free (name);

  /* link the RTP pad to the session manager */
  gst_pad_link (stream->srcpad, stream->send_rtp_sink);

  /* link udp elements */
  pad = gst_element_get_static_pad (stream->udpsink[0], "sink");
  gst_pad_link (stream->send_rtp_src, pad);
  gst_object_unref (pad);
  pad = gst_element_get_static_pad (stream->udpsink[1], "sink");
  gst_pad_link (stream->send_rtcp_src, pad);
  gst_object_unref (pad);
  pad = gst_element_get_static_pad (stream->udpsrc[1], "src");
  gst_pad_link (pad, stream->recv_rtcp_sink);
  gst_object_unref (pad);

  /* we set and keep these to playing so that they don't cause NO_PREROLL return
   * values */
  gst_element_set_state (stream->udpsrc[0], GST_STATE_PLAYING);
  gst_element_set_state (stream->udpsrc[1], GST_STATE_PLAYING);
  gst_element_set_locked_state (stream->udpsrc[0], TRUE);
  gst_element_set_locked_state (stream->udpsrc[1], TRUE);
 
  /* be notified of caps changes */
  stream->caps_sig = g_signal_connect (stream->send_rtp_sink, "notify::caps",
                  (GCallback) caps_notify, stream);

  stream->prepared = TRUE;

  return TRUE;
}


/**
 * gst_rtsp_media_prepare:
 * @obj: a #GstRTSPMedia
 *
 * Prepare @media for streaming. This function will create the pipeline and
 * other objects to manage the streaming.
 *
 * Returns: %TRUE on success.
 */
gboolean
gst_rtsp_media_prepare (GstRTSPMedia *media)
{
  GstStateChangeReturn ret;
  guint i, n_streams;

  if (media->prepared)
    goto was_prepared;

  media->pipeline = gst_pipeline_new ("media-pipeline");

  gst_bin_add (GST_BIN_CAST (media->pipeline), media->element);

  media->rtpbin = gst_element_factory_make ("gstrtpbin", "rtpbin");

  /* add stuf to the bin */
  gst_bin_add (GST_BIN (media->pipeline), media->rtpbin);

  ret = gst_element_set_state (media->pipeline, GST_STATE_READY);

  n_streams = gst_rtsp_media_n_streams (media);
  for (i = 0; i < n_streams; i++) {
    GstRTSPMediaStream *stream;

    stream = gst_rtsp_media_get_stream (media, i);

    setup_stream (stream, media);
  }

  /* first go to PAUSED */
  ret = gst_element_set_state (media->pipeline, GST_STATE_PAUSED);

  switch (ret) {
    case GST_STATE_CHANGE_SUCCESS:
      break;
    case GST_STATE_CHANGE_ASYNC:
      break;
    case GST_STATE_CHANGE_NO_PREROLL:
      /* we need to go to PLAYING */
      g_message ("live media %p", media);
      ret = gst_element_set_state (media->pipeline, GST_STATE_PLAYING);
      break;
    case GST_STATE_CHANGE_FAILURE:
      goto state_failed;
  }

  /* no wait for all pads to be prerolled */
  ret = gst_element_get_state (media->pipeline, NULL, NULL, -1);

  /* and back to PAUSED for live pipelines */
  ret = gst_element_set_state (media->pipeline, GST_STATE_PAUSED);

  g_message ("object %p is prerolled", media);
  media->prepared = TRUE;

  return TRUE;

  /* OK */
was_prepared:
  {
    return TRUE;
  }
  /* ERRORS */
state_failed:
  {
    g_message ("state change failed for media %p", media);
    return FALSE;
  }
}

gboolean
gst_rtsp_media_stream_add (GstRTSPMediaStream *stream, GstRTSPTransport *ct)
{
  g_return_val_if_fail (stream != NULL, FALSE);
  g_return_val_if_fail (ct != NULL, FALSE);
  g_return_val_if_fail (stream->prepared, FALSE);

  g_message ("adding %s:%d", ct->destination, ct->client_port.min);

  g_signal_emit_by_name (stream->udpsink[0], "add", ct->destination, ct->client_port.min, NULL);
  g_signal_emit_by_name (stream->udpsink[1], "add", ct->destination, ct->client_port.max, NULL);

  return TRUE;
}

gboolean
gst_rtsp_media_stream_remove (GstRTSPMediaStream *stream, GstRTSPTransport *ct)
{
  g_return_val_if_fail (stream != NULL, FALSE);
  g_return_val_if_fail (ct != NULL, FALSE);
  g_return_val_if_fail (stream->prepared, FALSE);

  g_message ("removing %s:%d", ct->destination, ct->client_port.min);

  g_signal_emit_by_name (stream->udpsink[0], "remove", ct->destination, ct->client_port.min, NULL);
  g_signal_emit_by_name (stream->udpsink[1], "remove", ct->destination, ct->client_port.max, NULL);

  return TRUE;
}

/**
 * gst_rtsp_media_play:
 * @media: a #GstRTSPMedia
 *
 * Tell the @media to start playing and streaming to the client.
 *
 * Returns: a #GstStateChangeReturn
 */
GstStateChangeReturn
gst_rtsp_media_play (GstRTSPMedia *media)
{
  GstStateChangeReturn ret;

  g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_STATE_CHANGE_FAILURE);
  g_return_val_if_fail (media->prepared, GST_STATE_CHANGE_FAILURE);

  g_message ("playing");
  ret = gst_element_set_state (media->pipeline, GST_STATE_PLAYING);

  return ret;
}

/**
 * gst_rtsp_media_pause:
 * @media: a #GstRTSPMedia
 *
 * Tell the @media to pause.
 *
 * Returns: a #GstStateChangeReturn
 */
GstStateChangeReturn
gst_rtsp_media_pause (GstRTSPMedia *media)
{
  GstStateChangeReturn ret;

  g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_STATE_CHANGE_FAILURE);
  g_return_val_if_fail (media->prepared, GST_STATE_CHANGE_FAILURE);

  g_message ("paused");
  ret = gst_element_set_state (media->pipeline, GST_STATE_PAUSED);

  return ret;
}

/**
 * gst_rtsp_media_stop:
 * @media: a #GstRTSPMedia
 *
 * Tell the @media to stop playing. After this call the media
 * cannot be played or paused anymore
 *
 * Returns: a #GstStateChangeReturn
 */
GstStateChangeReturn
gst_rtsp_media_stop (GstRTSPMedia *media)
{
  GstStateChangeReturn ret;

  g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_STATE_CHANGE_FAILURE);
  g_return_val_if_fail (media->prepared, GST_STATE_CHANGE_FAILURE);

  g_message ("stop");
  ret = gst_element_set_state (media->pipeline, GST_STATE_NULL);

  return ret;
}