/* GStreamer * Copyright (C) 2008 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., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, USA. */ /** * SECTION:rtsp-session-media * @short_description: Media managed in a session * @see_also: #GstRTSPMedia, #GstRTSPSession * * The #GstRTSPSessionMedia object manages a #GstRTSPMedia with a given path. * * With gst_rtsp_session_media_get_transport() and * gst_rtsp_session_media_set_transport() the transports of a #GstRTSPStream of * the managed #GstRTSPMedia can be retrieved and configured. * * Use gst_rtsp_session_media_set_state() to control the media state and * transports. * * Last reviewed on 2013-07-16 (1.0.0) */ #include #include "rtsp-session.h" #define GST_RTSP_SESSION_MEDIA_GET_PRIVATE(obj) \ (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_SESSION_MEDIA, GstRTSPSessionMediaPrivate)) struct _GstRTSPSessionMediaPrivate { GMutex lock; gchar *path; /* unmutable */ gint path_len; /* unmutable */ GstRTSPMedia *media; /* unmutable */ GstRTSPState state; /* protected by lock */ guint counter; /* protected by lock */ GPtrArray *transports; /* protected by lock */ }; enum { PROP_0, PROP_LAST }; GST_DEBUG_CATEGORY_STATIC (rtsp_session_media_debug); #define GST_CAT_DEFAULT rtsp_session_media_debug static void gst_rtsp_session_media_finalize (GObject * obj); G_DEFINE_TYPE (GstRTSPSessionMedia, gst_rtsp_session_media, G_TYPE_OBJECT); static void gst_rtsp_session_media_class_init (GstRTSPSessionMediaClass * klass) { GObjectClass *gobject_class; g_type_class_add_private (klass, sizeof (GstRTSPSessionMediaPrivate)); gobject_class = G_OBJECT_CLASS (klass); gobject_class->finalize = gst_rtsp_session_media_finalize; GST_DEBUG_CATEGORY_INIT (rtsp_session_media_debug, "rtspsessionmedia", 0, "GstRTSPSessionMedia"); } static void gst_rtsp_session_media_init (GstRTSPSessionMedia * media) { GstRTSPSessionMediaPrivate *priv = GST_RTSP_SESSION_MEDIA_GET_PRIVATE (media); media->priv = priv; g_mutex_init (&priv->lock); priv->state = GST_RTSP_STATE_INIT; } static void gst_rtsp_session_media_finalize (GObject * obj) { GstRTSPSessionMedia *media; GstRTSPSessionMediaPrivate *priv; media = GST_RTSP_SESSION_MEDIA (obj); priv = media->priv; GST_INFO ("free session media %p", media); gst_rtsp_session_media_set_state (media, GST_STATE_NULL); g_ptr_array_unref (priv->transports); g_free (priv->path); g_object_unref (priv->media); g_mutex_clear (&priv->lock); G_OBJECT_CLASS (gst_rtsp_session_media_parent_class)->finalize (obj); } static void free_session_media (gpointer data) { if (data) g_object_unref (data); } /** * gst_rtsp_session_media_new: * @path: the path * @media: (transfer full): the #GstRTSPMedia * * Create a new #GstRTSPSessionMedia that manages the streams * in @media for @path. @media should be prepared. * * Ownership is taken of @media. * * Returns: (transfer full): a new #GstRTSPSessionMedia. */ GstRTSPSessionMedia * gst_rtsp_session_media_new (const gchar * path, GstRTSPMedia * media) { GstRTSPSessionMediaPrivate *priv; GstRTSPSessionMedia *result; guint n_streams; GstRTSPMediaStatus status; g_return_val_if_fail (path != NULL, NULL); g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL); status = gst_rtsp_media_get_status (media); g_return_val_if_fail (status == GST_RTSP_MEDIA_STATUS_PREPARED || status == GST_RTSP_MEDIA_STATUS_SUSPENDED, NULL); result = g_object_new (GST_TYPE_RTSP_SESSION_MEDIA, NULL); priv = result->priv; priv->path = g_strdup (path); priv->path_len = strlen (path); priv->media = media; /* prealloc the streams now, filled with NULL */ n_streams = gst_rtsp_media_n_streams (media); priv->transports = g_ptr_array_new_full (n_streams, free_session_media); g_ptr_array_set_size (priv->transports, n_streams); return result; } /** * gst_rtsp_session_media_matches: * @media: a #GstRTSPSessionMedia * @path: a path * @matched: (out): the amount of matched characters of @path * * Check if the path of @media matches @path. It @path matches, the amount of * matched characters is returned in @matched. * * Returns: %TRUE when @path matches the path of @media. */ gboolean gst_rtsp_session_media_matches (GstRTSPSessionMedia * media, const gchar * path, gint * matched) { GstRTSPSessionMediaPrivate *priv; gint len; g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), FALSE); g_return_val_if_fail (path != NULL, FALSE); g_return_val_if_fail (matched != NULL, FALSE); priv = media->priv; len = strlen (path); /* path needs to be smaller than the media path */ if (len < priv->path_len) return FALSE; /* if media path is larger, it there should be a / following the path */ if (len > priv->path_len && path[priv->path_len] != '/') return FALSE; *matched = priv->path_len; return strncmp (path, priv->path, priv->path_len) == 0; } /** * gst_rtsp_session_media_get_media: * @media: a #GstRTSPSessionMedia * * Get the #GstRTSPMedia that was used when constructing @media * * Returns: (transfer none): the #GstRTSPMedia of @media. Remains valid as long * as @media is valid. */ GstRTSPMedia * gst_rtsp_session_media_get_media (GstRTSPSessionMedia * media) { g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), NULL); return media->priv->media; } /** * gst_rtsp_session_media_get_base_time: * @media: a #GstRTSPSessionMedia * * Get the base_time of the #GstRTSPMedia in @media * * Returns: the base_time of the media. */ GstClockTime gst_rtsp_session_media_get_base_time (GstRTSPSessionMedia * media) { g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), GST_CLOCK_TIME_NONE); return gst_rtsp_media_get_base_time (media->priv->media); } /** * gst_rtsp_session_media_get_rtpinfo: * @media: a #GstRTSPSessionMedia * * Retrieve the RTP-Info header string for all streams in @media * with configured transports. * * Returns: (transfer full): The RTP-Info as a string or %NULL when * no RTP-Info could be generated, g_free() after usage. */ gchar * gst_rtsp_session_media_get_rtpinfo (GstRTSPSessionMedia * media) { GstRTSPSessionMediaPrivate *priv; GString *rtpinfo = NULL; GstRTSPStreamTransport *transport; GstRTSPStream *stream; guint i, n_streams; GstClockTime earliest = GST_CLOCK_TIME_NONE; g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), NULL); priv = media->priv; g_mutex_lock (&priv->lock); if (gst_rtsp_media_get_status (priv->media) != GST_RTSP_MEDIA_STATUS_PREPARED) goto not_prepared; n_streams = priv->transports->len; /* first step, take lowest running-time from all streams */ GST_LOG_OBJECT (media, "determining start time among %d transports", n_streams); for (i = 0; i < n_streams; i++) { GstClockTime running_time; transport = g_ptr_array_index (priv->transports, i); if (transport == NULL) { GST_DEBUG_OBJECT (media, "ignoring unconfigured transport %d", i); continue; } stream = gst_rtsp_stream_transport_get_stream (transport); if (!gst_rtsp_stream_get_rtpinfo (stream, NULL, NULL, NULL, &running_time)) continue; GST_LOG_OBJECT (media, "running time of %d stream: %" GST_TIME_FORMAT, i, GST_TIME_ARGS (running_time)); if (!GST_CLOCK_TIME_IS_VALID (earliest)) { earliest = running_time; } else { earliest = MIN (earliest, running_time); } } GST_LOG_OBJECT (media, "media start time: %" GST_TIME_FORMAT, GST_TIME_ARGS (earliest)); /* next step, scale all rtptime of all streams to lowest running-time */ GST_LOG_OBJECT (media, "collecting RTP info for %d transports", n_streams); for (i = 0; i < n_streams; i++) { gchar *stream_rtpinfo; transport = g_ptr_array_index (priv->transports, i); if (transport == NULL) { GST_DEBUG_OBJECT (media, "ignoring unconfigured transport %d", i); continue; } stream_rtpinfo = gst_rtsp_stream_transport_get_rtpinfo (transport, earliest); if (stream_rtpinfo == NULL) { GST_DEBUG_OBJECT (media, "ignoring unknown RTPInfo %d", i); continue; } if (rtpinfo == NULL) rtpinfo = g_string_new (""); else g_string_append (rtpinfo, ", "); g_string_append (rtpinfo, stream_rtpinfo); g_free (stream_rtpinfo); } g_mutex_unlock (&priv->lock); if (rtpinfo == NULL) { GST_WARNING_OBJECT (media, "RTP info is empty"); return NULL; } return g_string_free (rtpinfo, FALSE); /* ERRORS */ not_prepared: { g_mutex_unlock (&priv->lock); GST_ERROR_OBJECT (media, "media was not prepared"); return NULL; } } /** * gst_rtsp_session_media_set_transport: * @media: a #GstRTSPSessionMedia * @stream: a #GstRTSPStream * @tr: (transfer full): a #GstRTSPTransport * * Configure the transport for @stream to @tr in @media. * * Returns: (transfer none): the new or updated #GstRTSPStreamTransport for @stream. */ GstRTSPStreamTransport * gst_rtsp_session_media_set_transport (GstRTSPSessionMedia * media, GstRTSPStream * stream, GstRTSPTransport * tr) { GstRTSPSessionMediaPrivate *priv; GstRTSPStreamTransport *result; guint idx; g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), NULL); g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL); g_return_val_if_fail (tr != NULL, NULL); priv = media->priv; idx = gst_rtsp_stream_get_index (stream); g_return_val_if_fail (idx < priv->transports->len, NULL); g_mutex_lock (&priv->lock); result = g_ptr_array_index (priv->transports, idx); if (result == NULL) { result = gst_rtsp_stream_transport_new (stream, tr); g_ptr_array_index (priv->transports, idx) = result; g_mutex_unlock (&priv->lock); } else { gst_rtsp_stream_transport_set_transport (result, tr); g_mutex_unlock (&priv->lock); } return result; } /** * gst_rtsp_session_media_get_transport: * @media: a #GstRTSPSessionMedia * @idx: the stream index * * Get a previously created #GstRTSPStreamTransport for the stream at @idx. * * Returns: (transfer none): a #GstRTSPStreamTransport that is valid until the * session of @media is unreffed. */ GstRTSPStreamTransport * gst_rtsp_session_media_get_transport (GstRTSPSessionMedia * media, guint idx) { GstRTSPSessionMediaPrivate *priv; GstRTSPStreamTransport *result; g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), NULL); priv = media->priv; g_return_val_if_fail (idx < priv->transports->len, NULL); g_mutex_lock (&priv->lock); result = g_ptr_array_index (priv->transports, idx); g_mutex_unlock (&priv->lock); return result; } /** * gst_rtsp_session_media_alloc_channels: * @media: a #GstRTSPSessionMedia * @range: (out): a #GstRTSPRange * * Fill @range with the next available min and max channels for * interleaved transport. * * Returns: %TRUE on success. */ gboolean gst_rtsp_session_media_alloc_channels (GstRTSPSessionMedia * media, GstRTSPRange * range) { GstRTSPSessionMediaPrivate *priv; g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), FALSE); priv = media->priv; g_mutex_lock (&priv->lock); range->min = priv->counter++; range->max = priv->counter++; g_mutex_unlock (&priv->lock); return TRUE; } /** * gst_rtsp_session_media_set_state: * @media: a #GstRTSPSessionMedia * @state: the new state * * Tell the media object @media to change to @state. * * Returns: %TRUE on success. */ gboolean gst_rtsp_session_media_set_state (GstRTSPSessionMedia * media, GstState state) { GstRTSPSessionMediaPrivate *priv; gboolean ret; g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), FALSE); priv = media->priv; g_mutex_lock (&priv->lock); ret = gst_rtsp_media_set_state (priv->media, state, priv->transports); g_mutex_unlock (&priv->lock); return ret; } /** * gst_rtsp_session_media_set_rtsp_state: * @media: a #GstRTSPSessionMedia * @state: a #GstRTSPState * * Set the RTSP state of @media to @state. */ void gst_rtsp_session_media_set_rtsp_state (GstRTSPSessionMedia * media, GstRTSPState state) { GstRTSPSessionMediaPrivate *priv; g_return_if_fail (GST_IS_RTSP_SESSION_MEDIA (media)); priv = media->priv; g_mutex_lock (&priv->lock); priv->state = state; g_mutex_unlock (&priv->lock); } /** * gst_rtsp_session_media_get_rtsp_state: * @media: a #GstRTSPSessionMedia * * Get the current RTSP state of @media. * * Returns: the current RTSP state of @media. */ GstRTSPState gst_rtsp_session_media_get_rtsp_state (GstRTSPSessionMedia * media) { GstRTSPSessionMediaPrivate *priv; GstRTSPState ret; g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), GST_RTSP_STATE_INVALID); priv = media->priv; g_mutex_lock (&priv->lock); ret = priv->state; g_mutex_unlock (&priv->lock); return ret; }