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8aaa432d58
And use it to correct the rtptime to the requested start-time. See https://bugzilla.gnome.org/show_bug.cgi?id=712198
512 lines
13 KiB
C
512 lines
13 KiB
C
/* GStreamer
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* Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:rtsp-session-media
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* @short_description: Media managed in a session
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* @see_also: #GstRTSPMedia, #GstRTSPSession
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*
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* The #GstRTSPSessionMedia object manages a #GstRTSPMedia with a given path.
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*
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* With gst_rtsp_session_media_get_transport() and
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* gst_rtsp_session_media_set_transport() the transports of a #GstRTSPStream of
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* the managed #GstRTSPMedia can be retrieved and configured.
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*
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* Use gst_rtsp_session_media_set_state() to control the media state and
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* transports.
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*
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* Last reviewed on 2013-07-16 (1.0.0)
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*/
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#include <string.h>
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#include "rtsp-session.h"
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#define GST_RTSP_SESSION_MEDIA_GET_PRIVATE(obj) \
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(G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_SESSION_MEDIA, GstRTSPSessionMediaPrivate))
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struct _GstRTSPSessionMediaPrivate
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{
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GMutex lock;
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gchar *path; /* unmutable */
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gint path_len; /* unmutable */
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GstRTSPMedia *media; /* unmutable */
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GstRTSPState state; /* protected by lock */
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guint counter; /* protected by lock */
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GPtrArray *transports; /* protected by lock */
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};
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enum
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{
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PROP_0,
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PROP_LAST
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};
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GST_DEBUG_CATEGORY_STATIC (rtsp_session_media_debug);
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#define GST_CAT_DEFAULT rtsp_session_media_debug
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static void gst_rtsp_session_media_finalize (GObject * obj);
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G_DEFINE_TYPE (GstRTSPSessionMedia, gst_rtsp_session_media, G_TYPE_OBJECT);
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static void
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gst_rtsp_session_media_class_init (GstRTSPSessionMediaClass * klass)
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{
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GObjectClass *gobject_class;
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g_type_class_add_private (klass, sizeof (GstRTSPSessionMediaPrivate));
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gobject_class = G_OBJECT_CLASS (klass);
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gobject_class->finalize = gst_rtsp_session_media_finalize;
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GST_DEBUG_CATEGORY_INIT (rtsp_session_media_debug, "rtspsessionmedia", 0,
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"GstRTSPSessionMedia");
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}
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static void
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gst_rtsp_session_media_init (GstRTSPSessionMedia * media)
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{
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GstRTSPSessionMediaPrivate *priv = GST_RTSP_SESSION_MEDIA_GET_PRIVATE (media);
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media->priv = priv;
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g_mutex_init (&priv->lock);
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priv->state = GST_RTSP_STATE_INIT;
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}
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static void
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gst_rtsp_session_media_finalize (GObject * obj)
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{
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GstRTSPSessionMedia *media;
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GstRTSPSessionMediaPrivate *priv;
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media = GST_RTSP_SESSION_MEDIA (obj);
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priv = media->priv;
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GST_INFO ("free session media %p", media);
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gst_rtsp_session_media_set_state (media, GST_STATE_NULL);
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g_ptr_array_unref (priv->transports);
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g_free (priv->path);
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g_object_unref (priv->media);
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g_mutex_clear (&priv->lock);
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G_OBJECT_CLASS (gst_rtsp_session_media_parent_class)->finalize (obj);
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}
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static void
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free_session_media (gpointer data)
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{
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if (data)
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g_object_unref (data);
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}
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/**
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* gst_rtsp_session_media_new:
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* @path: the path
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* @media: the #GstRTSPMedia
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*
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* Create a new #GstRTSPSessionMedia that manages the streams
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* in @media for @path. @media should be prepared.
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*
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* Ownership is taken of @media.
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*
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* Returns: a new #GstRTSPSessionMedia.
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*/
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GstRTSPSessionMedia *
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gst_rtsp_session_media_new (const gchar * path, GstRTSPMedia * media)
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{
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GstRTSPSessionMediaPrivate *priv;
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GstRTSPSessionMedia *result;
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guint n_streams;
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GstRTSPMediaStatus status;
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g_return_val_if_fail (path != NULL, NULL);
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g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
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status = gst_rtsp_media_get_status (media);
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g_return_val_if_fail (status == GST_RTSP_MEDIA_STATUS_PREPARED || status ==
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GST_RTSP_MEDIA_STATUS_SUSPENDED, NULL);
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result = g_object_new (GST_TYPE_RTSP_SESSION_MEDIA, NULL);
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priv = result->priv;
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priv->path = g_strdup (path);
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priv->path_len = strlen (path);
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priv->media = media;
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/* prealloc the streams now, filled with NULL */
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n_streams = gst_rtsp_media_n_streams (media);
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priv->transports = g_ptr_array_new_full (n_streams, free_session_media);
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g_ptr_array_set_size (priv->transports, n_streams);
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return result;
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}
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/**
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* gst_rtsp_session_media_matches:
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* @media: a #GstRTSPSessionMedia
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* @path: a path
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* @matched: (out): the amount of matched characters of @path
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*
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* Check if the path of @media matches @path. It @path matches, the amount of
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* matched characters is returned in @matched.
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*
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* Returns: %TRUE when @path matches the path of @media.
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*/
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gboolean
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gst_rtsp_session_media_matches (GstRTSPSessionMedia * media,
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const gchar * path, gint * matched)
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{
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GstRTSPSessionMediaPrivate *priv;
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gint len;
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g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), FALSE);
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g_return_val_if_fail (path != NULL, FALSE);
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g_return_val_if_fail (matched != NULL, FALSE);
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priv = media->priv;
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len = strlen (path);
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/* path needs to be smaller than the media path */
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if (len < priv->path_len)
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return FALSE;
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/* if media path is larger, it there should be a / following the path */
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if (len > priv->path_len && path[priv->path_len] != '/')
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return FALSE;
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*matched = priv->path_len;
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return strncmp (path, priv->path, priv->path_len) == 0;
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}
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/**
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* gst_rtsp_session_media_get_media:
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* @media: a #GstRTSPSessionMedia
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*
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* Get the #GstRTSPMedia that was used when constructing @media
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*
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* Returns: (transfer none): the #GstRTSPMedia of @media. Remains valid as long
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* as @media is valid.
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*/
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GstRTSPMedia *
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gst_rtsp_session_media_get_media (GstRTSPSessionMedia * media)
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{
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g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), NULL);
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return media->priv->media;
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}
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/**
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* gst_rtsp_session_media_get_base_time:
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* @media: a #GstRTSPSessionMedia
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*
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* Get the base_time of the #GstRTSPMedia in @media
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*
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* Returns: the base_time of the media.
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*/
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GstClockTime
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gst_rtsp_session_media_get_base_time (GstRTSPSessionMedia * media)
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{
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g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), GST_CLOCK_TIME_NONE);
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return gst_rtsp_media_get_base_time (media->priv->media);
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}
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/**
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* gst_rtsp_session_media_get_rtpinfo:
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* @media: a #GstRTSPSessionMedia
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*
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* Retrieve the RTP-Info header string for all streams in @media
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* with configured transports.
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*
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* Returns: (transfer full): The RTP-Info as a string, g_free()
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* after usage.
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*/
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gchar *
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gst_rtsp_session_media_get_rtpinfo (GstRTSPSessionMedia * media)
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{
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GstRTSPSessionMediaPrivate *priv;
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GString *rtpinfo = NULL;
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GstRTSPStreamTransport *transport;
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GstRTSPStream *stream;
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guint i, n_streams;
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GstClockTime earliest = GST_CLOCK_TIME_NONE;
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g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), NULL);
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priv = media->priv;
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g_mutex_lock (&priv->lock);
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if (gst_rtsp_media_get_status (priv->media) != GST_RTSP_MEDIA_STATUS_PREPARED)
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goto not_prepared;
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n_streams = priv->transports->len;
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/* first step, take lowest running-time from all streams */
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GST_LOG_OBJECT (media, "determining start time among %d transports",
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n_streams);
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for (i = 0; i < n_streams; i++) {
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GstClockTime running_time;
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transport = g_ptr_array_index (priv->transports, i);
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if (transport == NULL) {
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GST_DEBUG_OBJECT (media, "ignoring unconfigured transport %d", i);
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continue;
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}
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stream = gst_rtsp_stream_transport_get_stream (transport);
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if (!gst_rtsp_stream_get_rtpinfo (stream, NULL, NULL, NULL, &running_time))
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continue;
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GST_LOG_OBJECT (media, "running time of %d stream: %" GST_TIME_FORMAT, i,
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GST_TIME_ARGS (running_time));
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if (!GST_CLOCK_TIME_IS_VALID (earliest)) {
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earliest = running_time;
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} else {
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earliest = MIN (earliest, running_time);
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}
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}
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GST_LOG_OBJECT (media, "media start time: %" GST_TIME_FORMAT,
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GST_TIME_ARGS (earliest));
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/* next step, scale all rtptime of all streams to lowest running-time */
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GST_LOG_OBJECT (media, "collecting RTP info for %d transports", n_streams);
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for (i = 0; i < n_streams; i++) {
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gchar *stream_rtpinfo;
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transport = g_ptr_array_index (priv->transports, i);
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if (transport == NULL) {
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GST_DEBUG_OBJECT (media, "ignoring unconfigured transport %d", i);
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continue;
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}
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stream_rtpinfo =
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gst_rtsp_stream_transport_get_rtpinfo (transport, earliest);
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if (stream_rtpinfo == NULL)
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goto stream_rtpinfo_missing;
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if (rtpinfo == NULL)
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rtpinfo = g_string_new ("");
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else
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g_string_append (rtpinfo, ", ");
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g_string_append (rtpinfo, stream_rtpinfo);
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g_free (stream_rtpinfo);
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}
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if (rtpinfo == NULL) {
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GST_INFO_OBJECT (media, "no transports configured, RTP info is empty");
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rtpinfo = g_string_new ("");
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}
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g_mutex_unlock (&priv->lock);
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return g_string_free (rtpinfo, FALSE);
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/* ERRORS */
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not_prepared:
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{
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g_mutex_unlock (&priv->lock);
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GST_ERROR_OBJECT (media, "media was not prepared");
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return NULL;
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}
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stream_rtpinfo_missing:
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{
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g_mutex_unlock (&priv->lock);
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g_string_free (rtpinfo, TRUE);
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GST_ERROR_OBJECT (media, "could not get stream %d rtpinfo", i);
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return NULL;
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}
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}
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/**
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* gst_rtsp_session_media_set_transport:
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* @media: a #GstRTSPSessionMedia
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* @stream: a #GstRTSPStream
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* @tr: a #GstRTSPTransport
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*
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* Configure the transport for @stream to @tr in @media.
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*
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* Returns: (transfer none): the new or updated #GstRTSPStreamTransport for @stream.
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*/
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GstRTSPStreamTransport *
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gst_rtsp_session_media_set_transport (GstRTSPSessionMedia * media,
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GstRTSPStream * stream, GstRTSPTransport * tr)
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{
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GstRTSPSessionMediaPrivate *priv;
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GstRTSPStreamTransport *result;
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guint idx;
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g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), NULL);
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g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
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g_return_val_if_fail (tr != NULL, NULL);
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priv = media->priv;
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idx = gst_rtsp_stream_get_index (stream);
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g_return_val_if_fail (idx < priv->transports->len, NULL);
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g_mutex_lock (&priv->lock);
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result = g_ptr_array_index (priv->transports, idx);
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if (result == NULL) {
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result = gst_rtsp_stream_transport_new (stream, tr);
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g_ptr_array_index (priv->transports, idx) = result;
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g_mutex_unlock (&priv->lock);
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} else {
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gst_rtsp_stream_transport_set_transport (result, tr);
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g_mutex_unlock (&priv->lock);
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}
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return result;
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}
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/**
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* gst_rtsp_session_media_get_transport:
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* @media: a #GstRTSPSessionMedia
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* @idx: the stream index
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*
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* Get a previously created #GstRTSPStreamTransport for the stream at @idx.
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*
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* Returns: (transfer none): a #GstRTSPStreamTransport that is valid until the
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* session of @media is unreffed.
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*/
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GstRTSPStreamTransport *
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gst_rtsp_session_media_get_transport (GstRTSPSessionMedia * media, guint idx)
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{
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GstRTSPSessionMediaPrivate *priv;
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GstRTSPStreamTransport *result;
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g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), NULL);
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priv = media->priv;
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g_return_val_if_fail (idx < priv->transports->len, NULL);
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g_mutex_lock (&priv->lock);
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result = g_ptr_array_index (priv->transports, idx);
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g_mutex_unlock (&priv->lock);
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return result;
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}
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/**
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* gst_rtsp_session_media_alloc_channels:
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* @media: a #GstRTSPSessionMedia
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* @range: a #GstRTSPRange
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*
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* Fill @range with the next available min and max channels for
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* interleaved transport.
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*
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* Returns: %TRUE on success.
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*/
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gboolean
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gst_rtsp_session_media_alloc_channels (GstRTSPSessionMedia * media,
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GstRTSPRange * range)
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{
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GstRTSPSessionMediaPrivate *priv;
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g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), FALSE);
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priv = media->priv;
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g_mutex_lock (&priv->lock);
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range->min = priv->counter++;
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range->max = priv->counter++;
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g_mutex_unlock (&priv->lock);
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return TRUE;
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}
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/**
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* gst_rtsp_session_media_set_state:
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* @media: a #GstRTSPSessionMedia
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* @state: the new state
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*
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* Tell the media object @media to change to @state.
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*
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* Returns: %TRUE on success.
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*/
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gboolean
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gst_rtsp_session_media_set_state (GstRTSPSessionMedia * media, GstState state)
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{
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GstRTSPSessionMediaPrivate *priv;
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gboolean ret;
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g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), FALSE);
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priv = media->priv;
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g_mutex_lock (&priv->lock);
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ret = gst_rtsp_media_set_state (priv->media, state, priv->transports);
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g_mutex_unlock (&priv->lock);
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return ret;
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}
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/**
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* gst_rtsp_session_media_set_rtsp_state:
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* @media: a #GstRTSPSessionMedia
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* @state: a #GstRTSPState
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*
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* Set the RTSP state of @media to @state.
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*/
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void
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gst_rtsp_session_media_set_rtsp_state (GstRTSPSessionMedia * media,
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GstRTSPState state)
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{
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GstRTSPSessionMediaPrivate *priv;
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g_return_if_fail (GST_IS_RTSP_SESSION_MEDIA (media));
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priv = media->priv;
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g_mutex_lock (&priv->lock);
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priv->state = state;
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g_mutex_unlock (&priv->lock);
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}
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/**
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* gst_rtsp_session_media_get_rtsp_state:
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* @media: a #GstRTSPSessionMedia
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*
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* Get the current RTSP state of @media.
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*
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* Returns: the current RTSP state of @media.
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*/
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GstRTSPState
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gst_rtsp_session_media_get_rtsp_state (GstRTSPSessionMedia * media)
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{
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GstRTSPSessionMediaPrivate *priv;
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GstRTSPState ret;
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|
|
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;
|
|
}
|