/* GStreamer
* Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ * Copyright (C) 2015 Centricular Ltd
+ * Author: Sebastian Dröge <sebastian@centricular.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
+/**
+ * SECTION:rtsp-media
+ * @short_description: The media pipeline
+ * @see_also: #GstRTSPMediaFactory, #GstRTSPStream, #GstRTSPSession,
+ * #GstRTSPSessionMedia
+ *
+ * a #GstRTSPMedia contains the complete GStreamer pipeline to manage the
+ * streaming to the clients. The actual data transfer is done by the
+ * #GstRTSPStream objects that are created and exposed by the #GstRTSPMedia.
+ *
+ * The #GstRTSPMedia is usually created from a #GstRTSPMediaFactory when the
+ * client does a DESCRIBE or SETUP of a resource.
+ *
+ * A media is created with gst_rtsp_media_new() that takes the element that will
+ * provide the streaming elements. For each of the streams, a new #GstRTSPStream
+ * object needs to be made with the gst_rtsp_media_create_stream() which takes
+ * the payloader element and the source pad that produces the RTP stream.
+ *
+ * The pipeline of the media is set to PAUSED with gst_rtsp_media_prepare(). The
+ * prepare method will add rtpbin and sinks and sources to send and receive RTP
+ * and RTCP packets from the clients. Each stream srcpad is connected to an
+ * input into the internal rtpbin.
+ *
+ * It is also possible to dynamically create #GstRTSPStream objects during the
+ * prepare phase. With gst_rtsp_media_get_status() you can check the status of
+ * the prepare phase.
+ *
+ * After the media is prepared, it is ready for streaming. It will usually be
+ * managed in a session with gst_rtsp_session_manage_media(). See
+ * #GstRTSPSession and #GstRTSPSessionMedia.
+ *
+ * The state of the media can be controlled with gst_rtsp_media_set_state ().
+ * Seeking can be done with gst_rtsp_media_seek().
+ *
+ * With gst_rtsp_media_unprepare() the pipeline is stopped and shut down. When
+ * gst_rtsp_media_set_eos_shutdown() an EOS will be sent to the pipeline to
+ * cleanly shut down.
+ *
+ * With gst_rtsp_media_set_shared(), the media can be shared between multiple
+ * clients. With gst_rtsp_media_set_reusable() you can control if the pipeline
+ * can be prepared again after an unprepare.
+ *
+ * Last reviewed on 2013-07-11 (1.0.0)
+ */
+#include <stdio.h>
#include <string.h>
#include <stdlib.h>
#include <gst/app/gstappsrc.h>
#include <gst/app/gstappsink.h>
+#include <gst/sdp/gstmikey.h>
+#include <gst/rtp/gstrtppayloads.h>
+
+#define AES_128_KEY_LEN 16
+#define AES_256_KEY_LEN 32
+
+#define HMAC_32_KEY_LEN 4
+#define HMAC_80_KEY_LEN 10
+
#include "rtsp-media.h"
#define GST_RTSP_MEDIA_GET_PRIVATE(obj) \
GCond cond;
/* protected by lock */
+ GstRTSPPermissions *permissions;
gboolean shared;
+ gboolean suspend_mode;
gboolean reusable;
+ GstRTSPProfile profiles;
GstRTSPLowerTrans protocols;
gboolean reused;
gboolean eos_shutdown;
guint buffer_size;
- GstRTSPAuth *auth;
GstRTSPAddressPool *pool;
+ gboolean blocked;
+ GstRTSPTransportMode transport_mode;
GstElement *element;
GRecMutex state_lock; /* locking order: state lock, lock */
GstElement *fakesink; /* protected by lock */
GSource *source;
guint id;
+ GstRTSPThread *thread;
gboolean time_provider;
GstNetTimeProvider *nettime;
GstRTSPTimeRange range; /* protected by lock */
GstClockTime range_start;
GstClockTime range_stop;
+
+ GList *payloads; /* protected by lock */
+ GstClockTime rtx_time; /* protected by lock */
+ guint latency; /* protected by lock */
};
#define DEFAULT_SHARED FALSE
+#define DEFAULT_SUSPEND_MODE GST_RTSP_SUSPEND_MODE_NONE
#define DEFAULT_REUSABLE FALSE
-#define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_TCP
-//#define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP_MCAST
+#define DEFAULT_PROFILES GST_RTSP_PROFILE_AVP
+#define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | \
+ GST_RTSP_LOWER_TRANS_TCP
#define DEFAULT_EOS_SHUTDOWN FALSE
#define DEFAULT_BUFFER_SIZE 0x80000
#define DEFAULT_TIME_PROVIDER FALSE
+#define DEFAULT_LATENCY 200
+#define DEFAULT_TRANSPORT_MODE GST_RTSP_TRANSPORT_MODE_PLAY
/* define to dump received RTCP packets */
#undef DUMP_STATS
{
PROP_0,
PROP_SHARED,
+ PROP_SUSPEND_MODE,
PROP_REUSABLE,
+ PROP_PROFILES,
PROP_PROTOCOLS,
PROP_EOS_SHUTDOWN,
PROP_BUFFER_SIZE,
PROP_ELEMENT,
PROP_TIME_PROVIDER,
+ PROP_LATENCY,
+ PROP_TRANSPORT_MODE,
PROP_LAST
};
enum
{
SIGNAL_NEW_STREAM,
+ SIGNAL_REMOVED_STREAM,
SIGNAL_PREPARED,
SIGNAL_UNPREPARED,
+ SIGNAL_TARGET_STATE,
SIGNAL_NEW_STATE,
SIGNAL_LAST
};
const GValue * value, GParamSpec * pspec);
static void gst_rtsp_media_finalize (GObject * obj);
-static gpointer do_loop (GstRTSPMediaClass * klass);
static gboolean default_handle_message (GstRTSPMedia * media,
GstMessage * message);
static void finish_unprepare (GstRTSPMedia * media);
+static gboolean default_prepare (GstRTSPMedia * media, GstRTSPThread * thread);
static gboolean default_unprepare (GstRTSPMedia * media);
+static gboolean default_suspend (GstRTSPMedia * media);
+static gboolean default_unsuspend (GstRTSPMedia * media);
+static gboolean default_convert_range (GstRTSPMedia * media,
+ GstRTSPTimeRange * range, GstRTSPRangeUnit unit);
+static gboolean default_query_position (GstRTSPMedia * media,
+ gint64 * position);
+static gboolean default_query_stop (GstRTSPMedia * media, gint64 * stop);
+static GstElement *default_create_rtpbin (GstRTSPMedia * media);
+static gboolean default_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
+ GstSDPInfo * info);
+static gboolean default_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp);
+
+static gboolean wait_preroll (GstRTSPMedia * media);
static guint gst_rtsp_media_signals[SIGNAL_LAST] = { 0 };
+#define C_ENUM(v) ((gint) v)
+
+GType
+gst_rtsp_suspend_mode_get_type (void)
+{
+ static gsize id = 0;
+ static const GEnumValue values[] = {
+ {C_ENUM (GST_RTSP_SUSPEND_MODE_NONE), "GST_RTSP_SUSPEND_MODE_NONE", "none"},
+ {C_ENUM (GST_RTSP_SUSPEND_MODE_PAUSE), "GST_RTSP_SUSPEND_MODE_PAUSE",
+ "pause"},
+ {C_ENUM (GST_RTSP_SUSPEND_MODE_RESET), "GST_RTSP_SUSPEND_MODE_RESET",
+ "reset"},
+ {0, NULL, NULL}
+ };
+
+ if (g_once_init_enter (&id)) {
+ GType tmp = g_enum_register_static ("GstRTSPSuspendMode", values);
+ g_once_init_leave (&id, tmp);
+ }
+ return (GType) id;
+}
+
+#define C_FLAGS(v) ((guint) v)
+
+GType
+gst_rtsp_transport_mode_get_type (void)
+{
+ static gsize id = 0;
+ static const GFlagsValue values[] = {
+ {C_FLAGS (GST_RTSP_TRANSPORT_MODE_PLAY), "GST_RTSP_TRANSPORT_MODE_PLAY",
+ "play"},
+ {C_FLAGS (GST_RTSP_TRANSPORT_MODE_RECORD), "GST_RTSP_TRANSPORT_MODE_RECORD",
+ "record"},
+ {0, NULL, NULL}
+ };
+
+ if (g_once_init_enter (&id)) {
+ GType tmp = g_flags_register_static ("GstRTSPTransportMode", values);
+ g_once_init_leave (&id, tmp);
+ }
+ return (GType) id;
+}
+
G_DEFINE_TYPE (GstRTSPMedia, gst_rtsp_media, G_TYPE_OBJECT);
static void
"If this media pipeline can be shared", DEFAULT_SHARED,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ g_object_class_install_property (gobject_class, PROP_SUSPEND_MODE,
+ g_param_spec_enum ("suspend-mode", "Suspend Mode",
+ "How to suspend the media in PAUSED", GST_TYPE_RTSP_SUSPEND_MODE,
+ DEFAULT_SUSPEND_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
g_object_class_install_property (gobject_class, PROP_REUSABLE,
g_param_spec_boolean ("reusable", "Reusable",
"If this media pipeline can be reused after an unprepare",
DEFAULT_REUSABLE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ g_object_class_install_property (gobject_class, PROP_PROFILES,
+ g_param_spec_flags ("profiles", "Profiles",
+ "Allowed transfer profiles", GST_TYPE_RTSP_PROFILE,
+ DEFAULT_PROFILES, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
g_param_spec_flags ("protocols", "Protocols",
"Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
"The GstBin to use for streaming the media", GST_TYPE_ELEMENT,
G_PARAM_CONSTRUCT_ONLY | G_PARAM_READWRITE));
- g_object_class_install_property (gobject_class, PROP_EOS_SHUTDOWN,
+ g_object_class_install_property (gobject_class, PROP_TIME_PROVIDER,
g_param_spec_boolean ("time-provider", "Time Provider",
"Use a NetTimeProvider for clients",
DEFAULT_TIME_PROVIDER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ g_object_class_install_property (gobject_class, PROP_LATENCY,
+ g_param_spec_uint ("latency", "Latency",
+ "Latency used for receiving media in milliseconds", 0, G_MAXUINT,
+ DEFAULT_BUFFER_SIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_TRANSPORT_MODE,
+ g_param_spec_flags ("transport-mode", "Transport Mode",
+ "If this media pipeline can be used for PLAY or RECORD",
+ GST_TYPE_RTSP_TRANSPORT_MODE, DEFAULT_TRANSPORT_MODE,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
gst_rtsp_media_signals[SIGNAL_NEW_STREAM] =
g_signal_new ("new-stream", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
G_STRUCT_OFFSET (GstRTSPMediaClass, new_stream), NULL, NULL,
g_cclosure_marshal_generic, G_TYPE_NONE, 1, GST_TYPE_RTSP_STREAM);
+ gst_rtsp_media_signals[SIGNAL_REMOVED_STREAM] =
+ g_signal_new ("removed-stream", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaClass, removed_stream),
+ NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
+ GST_TYPE_RTSP_STREAM);
+
gst_rtsp_media_signals[SIGNAL_PREPARED] =
g_signal_new ("prepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
G_STRUCT_OFFSET (GstRTSPMediaClass, prepared), NULL, NULL,
- g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
+ g_cclosure_marshal_generic, G_TYPE_NONE, 0, G_TYPE_NONE);
gst_rtsp_media_signals[SIGNAL_UNPREPARED] =
g_signal_new ("unprepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
G_STRUCT_OFFSET (GstRTSPMediaClass, unprepared), NULL, NULL,
- g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
+ g_cclosure_marshal_generic, G_TYPE_NONE, 0, G_TYPE_NONE);
+
+ gst_rtsp_media_signals[SIGNAL_TARGET_STATE] =
+ g_signal_new ("target-state", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaClass, target_state),
+ NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1, G_TYPE_INT);
gst_rtsp_media_signals[SIGNAL_NEW_STATE] =
g_signal_new ("new-state", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
G_STRUCT_OFFSET (GstRTSPMediaClass, new_state), NULL, NULL,
- g_cclosure_marshal_VOID__INT, G_TYPE_NONE, 0, G_TYPE_INT);
-
- klass->context = g_main_context_new ();
- klass->loop = g_main_loop_new (klass->context, TRUE);
+ g_cclosure_marshal_generic, G_TYPE_NONE, 1, G_TYPE_INT);
GST_DEBUG_CATEGORY_INIT (rtsp_media_debug, "rtspmedia", 0, "GstRTSPMedia");
- klass->thread = g_thread_new ("Bus Thread", (GThreadFunc) do_loop, klass);
-
klass->handle_message = default_handle_message;
+ klass->prepare = default_prepare;
klass->unprepare = default_unprepare;
+ klass->suspend = default_suspend;
+ klass->unsuspend = default_unsuspend;
+ klass->convert_range = default_convert_range;
+ klass->query_position = default_query_position;
+ klass->query_stop = default_query_stop;
+ klass->create_rtpbin = default_create_rtpbin;
+ klass->setup_sdp = default_setup_sdp;
+ klass->handle_sdp = default_handle_sdp;
}
static void
g_rec_mutex_init (&priv->state_lock);
priv->shared = DEFAULT_SHARED;
+ priv->suspend_mode = DEFAULT_SUSPEND_MODE;
priv->reusable = DEFAULT_REUSABLE;
+ priv->profiles = DEFAULT_PROFILES;
priv->protocols = DEFAULT_PROTOCOLS;
priv->eos_shutdown = DEFAULT_EOS_SHUTDOWN;
priv->buffer_size = DEFAULT_BUFFER_SIZE;
priv->time_provider = DEFAULT_TIME_PROVIDER;
+ priv->transport_mode = DEFAULT_TRANSPORT_MODE;
}
static void
GST_INFO ("finalize media %p", media);
+ if (priv->permissions)
+ gst_rtsp_permissions_unref (priv->permissions);
+
g_ptr_array_unref (priv->streams);
g_list_free_full (priv->dynamic, gst_object_unref);
if (priv->nettime)
gst_object_unref (priv->nettime);
gst_object_unref (priv->element);
- if (priv->auth)
- g_object_unref (priv->auth);
if (priv->pool)
g_object_unref (priv->pool);
+ if (priv->payloads)
+ g_list_free (priv->payloads);
g_mutex_clear (&priv->lock);
g_cond_clear (&priv->cond);
g_rec_mutex_clear (&priv->state_lock);
case PROP_SHARED:
g_value_set_boolean (value, gst_rtsp_media_is_shared (media));
break;
+ case PROP_SUSPEND_MODE:
+ g_value_set_enum (value, gst_rtsp_media_get_suspend_mode (media));
+ break;
case PROP_REUSABLE:
g_value_set_boolean (value, gst_rtsp_media_is_reusable (media));
break;
+ case PROP_PROFILES:
+ g_value_set_flags (value, gst_rtsp_media_get_profiles (media));
+ break;
case PROP_PROTOCOLS:
g_value_set_flags (value, gst_rtsp_media_get_protocols (media));
break;
case PROP_TIME_PROVIDER:
g_value_set_boolean (value, gst_rtsp_media_is_time_provider (media));
break;
+ case PROP_LATENCY:
+ g_value_set_uint (value, gst_rtsp_media_get_latency (media));
+ break;
+ case PROP_TRANSPORT_MODE:
+ g_value_set_flags (value, gst_rtsp_media_get_transport_mode (media));
+ break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
}
switch (propid) {
case PROP_ELEMENT:
media->priv->element = g_value_get_object (value);
+ gst_object_ref_sink (media->priv->element);
break;
case PROP_SHARED:
gst_rtsp_media_set_shared (media, g_value_get_boolean (value));
break;
+ case PROP_SUSPEND_MODE:
+ gst_rtsp_media_set_suspend_mode (media, g_value_get_enum (value));
+ break;
case PROP_REUSABLE:
gst_rtsp_media_set_reusable (media, g_value_get_boolean (value));
break;
+ case PROP_PROFILES:
+ gst_rtsp_media_set_profiles (media, g_value_get_flags (value));
+ break;
case PROP_PROTOCOLS:
gst_rtsp_media_set_protocols (media, g_value_get_flags (value));
break;
case PROP_TIME_PROVIDER:
gst_rtsp_media_use_time_provider (media, g_value_get_boolean (value));
break;
+ case PROP_LATENCY:
+ gst_rtsp_media_set_latency (media, g_value_get_uint (value));
+ break;
+ case PROP_TRANSPORT_MODE:
+ gst_rtsp_media_set_transport_mode (media, g_value_get_flags (value));
+ break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
}
}
-static gpointer
-do_loop (GstRTSPMediaClass * klass)
+typedef struct
{
- GST_INFO ("enter mainloop");
- g_main_loop_run (klass->loop);
- GST_INFO ("exit mainloop");
+ gint64 position;
+ gboolean ret;
+} DoQueryPositionData;
- return NULL;
+static void
+do_query_position (GstRTSPStream * stream, DoQueryPositionData * data)
+{
+ gint64 tmp;
+
+ if (gst_rtsp_stream_query_position (stream, &tmp)) {
+ data->position = MAX (data->position, tmp);
+ data->ret = TRUE;
+ }
+}
+
+static gboolean
+default_query_position (GstRTSPMedia * media, gint64 * position)
+{
+ GstRTSPMediaPrivate *priv;
+ DoQueryPositionData data;
+
+ priv = media->priv;
+
+ data.position = -1;
+ data.ret = FALSE;
+
+ g_ptr_array_foreach (priv->streams, (GFunc) do_query_position, &data);
+
+ *position = data.position;
+
+ return data.ret;
+}
+
+typedef struct
+{
+ gint64 stop;
+ gboolean ret;
+} DoQueryStopData;
+
+static void
+do_query_stop (GstRTSPStream * stream, DoQueryStopData * data)
+{
+ gint64 tmp;
+
+ if (gst_rtsp_stream_query_stop (stream, &tmp)) {
+ data->stop = MAX (data->stop, tmp);
+ data->ret = TRUE;
+ }
+}
+
+static gboolean
+default_query_stop (GstRTSPMedia * media, gint64 * stop)
+{
+ GstRTSPMediaPrivate *priv;
+ DoQueryStopData data;
+
+ priv = media->priv;
+
+ data.stop = -1;
+ data.ret = FALSE;
+
+ g_ptr_array_foreach (priv->streams, (GFunc) do_query_stop, &data);
+
+ *stop = data.stop;
+
+ return data.ret;
+}
+
+static GstElement *
+default_create_rtpbin (GstRTSPMedia * media)
+{
+ GstElement *rtpbin;
+
+ rtpbin = gst_element_factory_make ("rtpbin", NULL);
+
+ return rtpbin;
}
/* must be called with state lock */
collect_media_stats (GstRTSPMedia * media)
{
GstRTSPMediaPrivate *priv = media->priv;
- gint64 position, duration;
+ gint64 position = 0, stop = -1;
if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
priv->range.max.seconds = -1;
priv->range_stop = -1;
} else {
+ GstRTSPMediaClass *klass;
+ gboolean ret;
+
+ klass = GST_RTSP_MEDIA_GET_CLASS (media);
+
/* get the position */
- if (!gst_element_query_position (priv->pipeline, GST_FORMAT_TIME,
- &position)) {
+ ret = FALSE;
+ if (klass->query_position)
+ ret = klass->query_position (media, &position);
+
+ if (!ret) {
GST_INFO ("position query failed");
position = 0;
}
- /* get the duration */
- if (!gst_element_query_duration (priv->pipeline, GST_FORMAT_TIME,
- &duration)) {
- GST_INFO ("duration query failed");
- duration = -1;
+ /* get the current segment stop */
+ ret = FALSE;
+ if (klass->query_stop)
+ ret = klass->query_stop (media, &stop);
+
+ if (!ret) {
+ GST_INFO ("stop query failed");
+ stop = -1;
}
- GST_INFO ("stats: position %" GST_TIME_FORMAT ", duration %"
- GST_TIME_FORMAT, GST_TIME_ARGS (position), GST_TIME_ARGS (duration));
+ GST_INFO ("stats: position %" GST_TIME_FORMAT ", stop %"
+ GST_TIME_FORMAT, GST_TIME_ARGS (position), GST_TIME_ARGS (stop));
if (position == -1) {
priv->range.min.type = GST_RTSP_TIME_NOW;
priv->range.min.seconds = ((gdouble) position) / GST_SECOND;
priv->range_start = position;
}
- if (duration == -1) {
+ if (stop == -1) {
priv->range.max.type = GST_RTSP_TIME_END;
priv->range.max.seconds = -1;
priv->range_stop = -1;
} else {
priv->range.max.type = GST_RTSP_TIME_SECONDS;
- priv->range.max.seconds = ((gdouble) duration) / GST_SECOND;
- priv->range_stop = duration;
+ priv->range.max.seconds = ((gdouble) stop) / GST_SECOND;
+ priv->range_stop = stop;
}
}
}
*
* Ownership is taken of @element.
*
- * Returns: a new #GstRTSPMedia object.
+ * Returns: (transfer full): a new #GstRTSPMedia object.
*/
GstRTSPMedia *
gst_rtsp_media_new (GstElement * element)
}
/**
- * gst_rtsp_media_take_element:
+ * gst_rtsp_media_get_element:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the element that was used when constructing @media.
+ *
+ * Returns: (transfer full): a #GstElement. Unref after usage.
+ */
+GstElement *
+gst_rtsp_media_get_element (GstRTSPMedia * media)
+{
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
+
+ return gst_object_ref (media->priv->element);
+}
+
+/**
+ * gst_rtsp_media_take_pipeline:
* @media: a #GstRTSPMedia
* @pipeline: (transfer full): a #GstPipeline
*
if (nettime)
gst_object_unref (nettime);
- gst_object_ref (priv->element);
gst_bin_add (GST_BIN_CAST (pipeline), priv->element);
}
/**
+ * gst_rtsp_media_set_permissions:
+ * @media: a #GstRTSPMedia
+ * @permissions: (transfer none): a #GstRTSPPermissions
+ *
+ * Set @permissions on @media.
+ */
+void
+gst_rtsp_media_set_permissions (GstRTSPMedia * media,
+ GstRTSPPermissions * permissions)
+{
+ GstRTSPMediaPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ if (priv->permissions)
+ gst_rtsp_permissions_unref (priv->permissions);
+ if ((priv->permissions = permissions))
+ gst_rtsp_permissions_ref (permissions);
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_get_permissions:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the permissions object from @media.
+ *
+ * Returns: (transfer full): a #GstRTSPPermissions object, unref after usage.
+ */
+GstRTSPPermissions *
+gst_rtsp_media_get_permissions (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ GstRTSPPermissions *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ if ((result = priv->permissions))
+ gst_rtsp_permissions_ref (result);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_media_set_suspend_mode:
+ * @media: a #GstRTSPMedia
+ * @mode: the new #GstRTSPSuspendMode
+ *
+ * Control how @ media will be suspended after the SDP has been generated and
+ * after a PAUSE request has been performed.
+ *
+ * Media must be unprepared when setting the suspend mode.
+ */
+void
+gst_rtsp_media_set_suspend_mode (GstRTSPMedia * media, GstRTSPSuspendMode mode)
+{
+ GstRTSPMediaPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+ priv = media->priv;
+
+ g_rec_mutex_lock (&priv->state_lock);
+ if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED)
+ goto was_prepared;
+ priv->suspend_mode = mode;
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ return;
+
+ /* ERRORS */
+was_prepared:
+ {
+ GST_WARNING ("media %p was prepared", media);
+ g_rec_mutex_unlock (&priv->state_lock);
+ }
+}
+
+/**
+ * gst_rtsp_media_get_suspend_mode:
+ * @media: a #GstRTSPMedia
+ *
+ * Get how @media will be suspended.
+ *
+ * Returns: #GstRTSPSuspendMode.
+ */
+GstRTSPSuspendMode
+gst_rtsp_media_get_suspend_mode (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ GstRTSPSuspendMode res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_RTSP_SUSPEND_MODE_NONE);
+
+ priv = media->priv;
+
+ g_rec_mutex_lock (&priv->state_lock);
+ res = priv->suspend_mode;
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ return res;
+}
+
+/**
* gst_rtsp_media_set_shared:
* @media: a #GstRTSPMedia
* @shared: the new value
return res;
}
+static void
+do_set_profiles (GstRTSPStream * stream, GstRTSPProfile * profiles)
+{
+ gst_rtsp_stream_set_profiles (stream, *profiles);
+}
+
+/**
+ * gst_rtsp_media_set_profiles:
+ * @media: a #GstRTSPMedia
+ * @profiles: the new flags
+ *
+ * Configure the allowed lower transport for @media.
+ */
+void
+gst_rtsp_media_set_profiles (GstRTSPMedia * media, GstRTSPProfile profiles)
+{
+ GstRTSPMediaPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->profiles = profiles;
+ g_ptr_array_foreach (priv->streams, (GFunc) do_set_profiles, &profiles);
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_get_profiles:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the allowed profiles of @media.
+ *
+ * Returns: a #GstRTSPProfile
+ */
+GstRTSPProfile
+gst_rtsp_media_get_profiles (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ GstRTSPProfile res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_RTSP_PROFILE_UNKNOWN);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ res = priv->profiles;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+static void
+do_set_protocols (GstRTSPStream * stream, GstRTSPLowerTrans * protocols)
+{
+ gst_rtsp_stream_set_protocols (stream, *protocols);
+}
+
/**
* gst_rtsp_media_set_protocols:
* @media: a #GstRTSPMedia
g_mutex_lock (&priv->lock);
priv->protocols = protocols;
+ g_ptr_array_foreach (priv->streams, (GFunc) do_set_protocols, &protocols);
g_mutex_unlock (&priv->lock);
}
}
/**
- * gst_rtsp_media_use_time_provider:
+ * gst_rtsp_media_set_retransmission_time:
* @media: a #GstRTSPMedia
+ * @time: the new value
*
- * Set @media to provide a GstNetTimeProvider.
+ * Set the amount of time to store retransmission packets.
*/
void
-gst_rtsp_media_use_time_provider (GstRTSPMedia * media, gboolean time_provider)
+gst_rtsp_media_set_retransmission_time (GstRTSPMedia * media, GstClockTime time)
{
GstRTSPMediaPrivate *priv;
+ guint i;
g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+ GST_LOG_OBJECT (media, "set retransmission time %" G_GUINT64_FORMAT, time);
+
priv = media->priv;
g_mutex_lock (&priv->lock);
- priv->time_provider = time_provider;
+ priv->rtx_time = time;
+ for (i = 0; i < priv->streams->len; i++) {
+ GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
+
+ gst_rtsp_stream_set_retransmission_time (stream, time);
+ }
+
+ if (priv->rtpbin)
+ g_object_set (priv->rtpbin, "do-retransmission", time > 0, NULL);
g_mutex_unlock (&priv->lock);
}
/**
- * gst_rtsp_media_is_time_provider:
+ * gst_rtsp_media_get_retransmission_time:
* @media: a #GstRTSPMedia
*
- * Check if @media can provide a #GstNetTimeProvider for its pipeline clock.
- *
- * Use gst_rtsp_media_get_time_provider() to get the network clock.
+ * Get the amount of time to store retransmission data.
*
- * Returns: %TRUE if @media can provide a #GstNetTimeProvider.
+ * Returns: the amount of time to store retransmission data.
*/
-gboolean
-gst_rtsp_media_is_time_provider (GstRTSPMedia * media)
+GstClockTime
+gst_rtsp_media_get_retransmission_time (GstRTSPMedia * media)
{
GstRTSPMediaPrivate *priv;
- gboolean res;
+ GstClockTime res;
g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
priv = media->priv;
g_mutex_unlock (&priv->lock);
- res = priv->time_provider;
+ res = priv->rtx_time;
g_mutex_unlock (&priv->lock);
return res;
}
/**
- * gst_rtsp_media_set_auth:
+ * gst_rtsp_media_set_latncy:
* @media: a #GstRTSPMedia
- * @auth: a #GstRTSPAuth
+ * @latency: latency in milliseconds
*
- * configure @auth to be used as the authentication manager of @media.
+ * Configure the latency used for receiving media.
*/
void
-gst_rtsp_media_set_auth (GstRTSPMedia * media, GstRTSPAuth * auth)
+gst_rtsp_media_set_latency (GstRTSPMedia * media, guint latency)
{
GstRTSPMediaPrivate *priv;
- GstRTSPAuth *old;
g_return_if_fail (GST_IS_RTSP_MEDIA (media));
- priv = media->priv;
+ GST_LOG_OBJECT (media, "set latency %ums", latency);
- GST_LOG_OBJECT (media, "set auth %p", auth);
+ priv = media->priv;
g_mutex_lock (&priv->lock);
- if ((old = priv->auth) != auth)
- priv->auth = auth ? g_object_ref (auth) : NULL;
- else
- old = NULL;
+ priv->latency = latency;
+ if (priv->rtpbin)
+ g_object_set (priv->rtpbin, "latency", latency, NULL);
g_mutex_unlock (&priv->lock);
-
- if (old)
- g_object_unref (old);
}
/**
- * gst_rtsp_media_get_auth:
+ * gst_rtsp_media_get_latency:
* @media: a #GstRTSPMedia
*
- * Get the #GstRTSPAuth used as the authentication manager of @media.
+ * Get the latency that is used for receiving media.
*
- * Returns: (transfer full): the #GstRTSPAuth of @media. g_object_unref() after
- * usage.
+ * Returns: latency in milliseconds
*/
-GstRTSPAuth *
-gst_rtsp_media_get_auth (GstRTSPMedia * media)
+guint
+gst_rtsp_media_get_latency (GstRTSPMedia * media)
{
GstRTSPMediaPrivate *priv;
- GstRTSPAuth *result;
+ guint res;
- g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+ priv = media->priv;
+
+ g_mutex_unlock (&priv->lock);
+ res = priv->latency;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_media_use_time_provider:
+ * @media: a #GstRTSPMedia
+ * @time_provider: if a #GstNetTimeProvider should be used
+ *
+ * Set @media to provide a #GstNetTimeProvider.
+ */
+void
+gst_rtsp_media_use_time_provider (GstRTSPMedia * media, gboolean time_provider)
+{
+ GstRTSPMediaPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
priv = media->priv;
g_mutex_lock (&priv->lock);
- if ((result = priv->auth))
- g_object_ref (result);
+ priv->time_provider = time_provider;
g_mutex_unlock (&priv->lock);
+}
- return result;
+/**
+ * gst_rtsp_media_is_time_provider:
+ * @media: a #GstRTSPMedia
+ *
+ * Check if @media can provide a #GstNetTimeProvider for its pipeline clock.
+ *
+ * Use gst_rtsp_media_get_time_provider() to get the network clock.
+ *
+ * Returns: %TRUE if @media can provide a #GstNetTimeProvider.
+ */
+gboolean
+gst_rtsp_media_is_time_provider (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ gboolean res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+ priv = media->priv;
+
+ g_mutex_unlock (&priv->lock);
+ res = priv->time_provider;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
}
/**
* gst_rtsp_media_set_address_pool:
* @media: a #GstRTSPMedia
- * @pool: a #GstRTSPAddressPool
+ * @pool: (transfer none): a #GstRTSPAddressPool
*
* configure @pool to be used as the address pool of @media.
*/
return result;
}
+static GList *
+_find_payload_types (GstRTSPMedia * media)
+{
+ gint i, n;
+ GQueue queue = G_QUEUE_INIT;
+
+ n = media->priv->streams->len;
+ for (i = 0; i < n; i++) {
+ GstRTSPStream *stream = g_ptr_array_index (media->priv->streams, i);
+ guint pt = gst_rtsp_stream_get_pt (stream);
+
+ g_queue_push_tail (&queue, GUINT_TO_POINTER (pt));
+ }
+
+ return queue.head;
+}
+
+static guint
+_next_available_pt (GList * payloads)
+{
+ guint i;
+
+ for (i = 96; i <= 127; i++) {
+ GList *iter = g_list_find (payloads, GINT_TO_POINTER (i));
+ if (!iter)
+ return GPOINTER_TO_UINT (i);
+ }
+
+ return 0;
+}
+
/**
* gst_rtsp_media_collect_streams:
* @media: a #GstRTSPMedia
*
- * Find all payloader elements, they should be named pay%d in the
+ * Find all payloader elements, they should be named pay\%d in the
* element of @media, and create #GstRTSPStreams for them.
*
- * Collect all dynamic elements, named dynpay%d, and add them to
+ * Collect all dynamic elements, named dynpay\%d, and add them to
* the list of dynamic elements.
+ *
+ * Find all depayloader elements, they should be named depay\%d in the
+ * element of @media, and create #GstRTSPStreams for them.
*/
void
gst_rtsp_media_collect_streams (GstRTSPMedia * media)
GstPad *pad;
gint i;
gboolean have_elem;
+ GstRTSPTransportMode mode = 0;
g_return_if_fail (GST_IS_RTSP_MEDIA (media));
gst_object_unref (elem);
have_elem = TRUE;
+ mode |= GST_RTSP_TRANSPORT_MODE_PLAY;
}
g_free (name);
name = g_strdup_printf ("dynpay%d", i);
if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
/* a stream that will dynamically create pads to provide RTP packets */
-
GST_INFO ("found dynamic element %d, %p", i, elem);
g_mutex_lock (&priv->lock);
g_mutex_unlock (&priv->lock);
have_elem = TRUE;
+ mode |= GST_RTSP_TRANSPORT_MODE_PLAY;
+ }
+ g_free (name);
+
+ name = g_strdup_printf ("depay%d", i);
+ if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
+ GST_INFO ("found stream %d with depayloader %p", i, elem);
+
+ /* take the pad of the payloader */
+ pad = gst_element_get_static_pad (elem, "sink");
+ /* create the stream */
+ gst_rtsp_media_create_stream (media, elem, pad);
+ gst_object_unref (pad);
+ gst_object_unref (elem);
+
+ have_elem = TRUE;
+ mode |= GST_RTSP_TRANSPORT_MODE_RECORD;
}
g_free (name);
}
+
+ if (have_elem) {
+ if (priv->transport_mode != mode)
+ GST_WARNING ("found different mode than expected (0x%02x != 0x%02d)",
+ priv->transport_mode, mode);
+ }
}
/**
* gst_rtsp_media_create_stream:
* @media: a #GstRTSPMedia
* @payloader: a #GstElement
- * @srcpad: a source #GstPad
+ * @pad: a #GstPad
*
- * Create a new stream in @media that provides RTP data on @srcpad.
- * @srcpad should be a pad of an element inside @media->element.
+ * Create a new stream in @media that provides RTP data on @pad.
+ * @pad should be a pad of an element inside @media->element.
*
* Returns: (transfer none): a new #GstRTSPStream that remains valid for as long
- * as @media exists.
+ * as @media exists.
*/
GstRTSPStream *
gst_rtsp_media_create_stream (GstRTSPMedia * media, GstElement * payloader,
{
GstRTSPMediaPrivate *priv;
GstRTSPStream *stream;
- GstPad *srcpad;
+ GstPad *ghostpad;
gchar *name;
gint idx;
g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
g_return_val_if_fail (GST_IS_PAD (pad), NULL);
- g_return_val_if_fail (GST_PAD_IS_SRC (pad), NULL);
priv = media->priv;
GST_DEBUG ("media %p: creating stream with index %d", media, idx);
- name = g_strdup_printf ("src_%u", idx);
- srcpad = gst_ghost_pad_new (name, pad);
- gst_pad_set_active (srcpad, TRUE);
- gst_element_add_pad (priv->element, srcpad);
+ if (GST_PAD_IS_SRC (pad))
+ name = g_strdup_printf ("src_%u", idx);
+ else
+ name = g_strdup_printf ("sink_%u", idx);
+
+ ghostpad = gst_ghost_pad_new (name, pad);
+ gst_pad_set_active (ghostpad, TRUE);
+ gst_element_add_pad (priv->element, ghostpad);
g_free (name);
- stream = gst_rtsp_stream_new (idx, payloader, srcpad);
+ stream = gst_rtsp_stream_new (idx, payloader, ghostpad);
if (priv->pool)
gst_rtsp_stream_set_address_pool (stream, priv->pool);
+ gst_rtsp_stream_set_profiles (stream, priv->profiles);
+ gst_rtsp_stream_set_protocols (stream, priv->protocols);
+ gst_rtsp_stream_set_retransmission_time (stream, priv->rtx_time);
g_ptr_array_add (priv->streams, stream);
+
+ if (GST_PAD_IS_SRC (pad)) {
+ gint i, n;
+
+ if (priv->payloads)
+ g_list_free (priv->payloads);
+ priv->payloads = _find_payload_types (media);
+
+ n = priv->streams->len;
+ for (i = 0; i < n; i++) {
+ GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
+ guint rtx_pt = _next_available_pt (priv->payloads);
+
+ if (rtx_pt == 0) {
+ GST_WARNING ("Ran out of space of dynamic payload types");
+ break;
+ }
+
+ gst_rtsp_stream_set_retransmission_pt (stream, rtx_pt);
+
+ priv->payloads =
+ g_list_append (priv->payloads, GUINT_TO_POINTER (rtx_pt));
+ }
+ }
g_mutex_unlock (&priv->lock);
g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STREAM], 0, stream,
return stream;
}
+static void
+gst_rtsp_media_remove_stream (GstRTSPMedia * media, GstRTSPStream * stream)
+{
+ GstRTSPMediaPrivate *priv;
+ GstPad *srcpad;
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ /* remove the ghostpad */
+ srcpad = gst_rtsp_stream_get_srcpad (stream);
+ gst_element_remove_pad (priv->element, srcpad);
+ gst_object_unref (srcpad);
+ /* now remove the stream */
+ g_object_ref (stream);
+ g_ptr_array_remove (priv->streams, stream);
+ g_mutex_unlock (&priv->lock);
+
+ g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_REMOVED_STREAM], 0,
+ stream, NULL);
+
+ g_object_unref (stream);
+}
+
/**
* gst_rtsp_media_n_streams:
* @media: a #GstRTSPMedia
*
* Retrieve the stream with index @idx from @media.
*
- * Returns: (transfer none): the #GstRTSPStream at index @idx or %NULL when a stream with
- * that index did not exist.
+ * Returns: (nullable) (transfer none): the #GstRTSPStream at index
+ * @idx or %NULL when a stream with that index did not exist.
*/
GstRTSPStream *
gst_rtsp_media_get_stream (GstRTSPMedia * media, guint idx)
}
/**
+ * gst_rtsp_media_find_stream:
+ * @media: a #GstRTSPMedia
+ * @control: the control of the stream
+ *
+ * Find a stream in @media with @control as the control uri.
+ *
+ * Returns: (nullable) (transfer none): the #GstRTSPStream with
+ * control uri @control or %NULL when a stream with that control did
+ * not exist.
+ */
+GstRTSPStream *
+gst_rtsp_media_find_stream (GstRTSPMedia * media, const gchar * control)
+{
+ GstRTSPMediaPrivate *priv;
+ GstRTSPStream *res;
+ gint i;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
+ g_return_val_if_fail (control != NULL, NULL);
+
+ priv = media->priv;
+
+ res = NULL;
+
+ g_mutex_lock (&priv->lock);
+ for (i = 0; i < priv->streams->len; i++) {
+ GstRTSPStream *test;
+
+ test = g_ptr_array_index (priv->streams, i);
+ if (gst_rtsp_stream_has_control (test, control)) {
+ res = test;
+ break;
+ }
+ }
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+/* called with state-lock */
+static gboolean
+default_convert_range (GstRTSPMedia * media, GstRTSPTimeRange * range,
+ GstRTSPRangeUnit unit)
+{
+ return gst_rtsp_range_convert_units (range, unit);
+}
+
+/**
* gst_rtsp_media_get_range_string:
* @media: a #GstRTSPMedia
* @play: for the PLAY request
* Get the current range as a string. @media must be prepared with
* gst_rtsp_media_prepare ().
*
- * Returns: The range as a string, g_free() after usage.
+ * Returns: (transfer full): The range as a string, g_free() after usage.
*/
gchar *
gst_rtsp_media_get_range_string (GstRTSPMedia * media, gboolean play,
GstRTSPRangeUnit unit)
{
+ GstRTSPMediaClass *klass;
GstRTSPMediaPrivate *priv;
gchar *result;
GstRTSPTimeRange range;
+ klass = GST_RTSP_MEDIA_GET_CLASS (media);
g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
+ g_return_val_if_fail (klass->convert_range != NULL, FALSE);
priv = media->priv;
g_rec_mutex_lock (&priv->state_lock);
- if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
+ if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
+ priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
goto not_prepared;
g_mutex_lock (&priv->lock);
+
+ /* Update the range value with current position/duration */
+ collect_media_stats (media);
+
/* make copy */
range = priv->range;
g_mutex_unlock (&priv->lock);
g_rec_mutex_unlock (&priv->state_lock);
- gst_rtsp_range_convert_units (&range, unit);
+ if (!klass->convert_range (media, &range, unit))
+ goto conversion_failed;
result = gst_rtsp_range_to_string (&range);
g_rec_mutex_unlock (&priv->state_lock);
return NULL;
}
+conversion_failed:
+ {
+ GST_WARNING ("range conversion to unit %d failed", unit);
+ return NULL;
+ }
+}
+
+static void
+stream_update_blocked (GstRTSPStream * stream, GstRTSPMedia * media)
+{
+ gst_rtsp_stream_set_blocked (stream, media->priv->blocked);
+}
+
+static void
+media_streams_set_blocked (GstRTSPMedia * media, gboolean blocked)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+
+ GST_DEBUG ("media %p set blocked %d", media, blocked);
+ priv->blocked = blocked;
+ g_ptr_array_foreach (priv->streams, (GFunc) stream_update_blocked, media);
+}
+
+static void
+gst_rtsp_media_set_status (GstRTSPMedia * media, GstRTSPMediaStatus status)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->status = status;
+ GST_DEBUG ("setting new status to %d", status);
+ g_cond_broadcast (&priv->cond);
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_get_status:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the status of @media. When @media is busy preparing, this function waits
+ * until @media is prepared or in error.
+ *
+ * Returns: the status of @media.
+ */
+GstRTSPMediaStatus
+gst_rtsp_media_get_status (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ GstRTSPMediaStatus result;
+ gint64 end_time;
+
+ g_mutex_lock (&priv->lock);
+ end_time = g_get_monotonic_time () + 20 * G_TIME_SPAN_SECOND;
+ /* while we are preparing, wait */
+ while (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING) {
+ GST_DEBUG ("waiting for status change");
+ if (!g_cond_wait_until (&priv->cond, &priv->lock, end_time)) {
+ GST_DEBUG ("timeout, assuming error status");
+ priv->status = GST_RTSP_MEDIA_STATUS_ERROR;
+ }
+ }
+ /* could be success or error */
+ result = priv->status;
+ GST_DEBUG ("got status %d", result);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
}
/**
* gst_rtsp_media_seek:
* @media: a #GstRTSPMedia
- * @range: a #GstRTSPTimeRange
+ * @range: (transfer none): a #GstRTSPTimeRange
*
* Seek the pipeline of @media to @range. @media must be prepared with
* gst_rtsp_media_prepare().
gboolean
gst_rtsp_media_seek (GstRTSPMedia * media, GstRTSPTimeRange * range)
{
+ GstRTSPMediaClass *klass;
GstRTSPMediaPrivate *priv;
- GstSeekFlags flags;
gboolean res;
GstClockTime start, stop;
GstSeekType start_type, stop_type;
+ GstQuery *query;
+
+ klass = GST_RTSP_MEDIA_GET_CLASS (media);
g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
g_return_val_if_fail (range != NULL, FALSE);
+ g_return_val_if_fail (klass->convert_range != NULL, FALSE);
priv = media->priv;
if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
goto not_prepared;
+ /* Update the seekable state of the pipeline in case it changed */
+ if ((priv->transport_mode & GST_RTSP_TRANSPORT_MODE_RECORD)) {
+ /* TODO: Seeking for RECORD? */
+ priv->seekable = FALSE;
+ } else {
+ query = gst_query_new_seeking (GST_FORMAT_TIME);
+ if (gst_element_query (priv->pipeline, query)) {
+ GstFormat format;
+ gboolean seekable;
+ gint64 start, end;
+
+ gst_query_parse_seeking (query, &format, &seekable, &start, &end);
+ priv->seekable = seekable;
+ }
+ gst_query_unref (query);
+ }
+
if (!priv->seekable)
goto not_seekable;
- /* depends on the current playing state of the pipeline. We might need to
- * queue this until we get EOS. */
- flags = GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_ACCURATE | GST_SEEK_FLAG_KEY_UNIT;
-
start_type = stop_type = GST_SEEK_TYPE_NONE;
- if (!gst_rtsp_range_get_times (range, &start, &stop))
+ if (!klass->convert_range (media, range, GST_RTSP_RANGE_NPT))
goto not_supported;
+ gst_rtsp_range_get_times (range, &start, &stop);
GST_INFO ("got %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
GST_INFO ("current %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
GST_TIME_ARGS (priv->range_start), GST_TIME_ARGS (priv->range_stop));
- if (priv->range_start == start)
- start = GST_CLOCK_TIME_NONE;
- else if (start != GST_CLOCK_TIME_NONE)
+ if (start != GST_CLOCK_TIME_NONE)
start_type = GST_SEEK_TYPE_SET;
if (priv->range_stop == stop)
stop_type = GST_SEEK_TYPE_SET;
if (start != GST_CLOCK_TIME_NONE || stop != GST_CLOCK_TIME_NONE) {
+ GstSeekFlags flags;
+
GST_INFO ("seeking to %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
+ if (priv->blocked)
+ media_streams_set_blocked (media, TRUE);
+
+ /* depends on the current playing state of the pipeline. We might need to
+ * queue this until we get EOS. */
+ flags = GST_SEEK_FLAG_FLUSH;
+
+ /* if range start was not supplied we must continue from current position.
+ * but since we're doing a flushing seek, let us query the current position
+ * so we end up at exactly the same position after the seek. */
+ if (range->min.type == GST_RTSP_TIME_END) { /* Yepp, that's right! */
+ gint64 position;
+ gboolean ret = FALSE;
+
+ if (klass->query_position)
+ ret = klass->query_position (media, &position);
+
+ if (!ret) {
+ GST_WARNING ("position query failed");
+ } else {
+ GST_DEBUG ("doing accurate seek to %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (position));
+ start = position;
+ start_type = GST_SEEK_TYPE_SET;
+ flags |= GST_SEEK_FLAG_ACCURATE;
+ }
+ } else {
+ /* only set keyframe flag when modifying start */
+ if (start_type != GST_SEEK_TYPE_NONE)
+ flags |= GST_SEEK_FLAG_KEY_UNIT;
+ }
+
+ /* FIXME, we only do forwards playback, no trick modes yet */
res = gst_element_seek (priv->pipeline, 1.0, GST_FORMAT_TIME,
flags, start_type, start, stop_type, stop);
/* and block for the seek to complete */
GST_INFO ("done seeking %d", res);
- gst_element_get_state (priv->pipeline, NULL, NULL, -1);
- GST_INFO ("prerolled again");
+ g_rec_mutex_unlock (&priv->state_lock);
- collect_media_stats (media);
+ /* wait until pipeline is prerolled again, this will also collect stats */
+ if (!wait_preroll (media))
+ goto preroll_failed;
+
+ g_rec_mutex_lock (&priv->state_lock);
+ GST_INFO ("prerolled again");
} else {
GST_INFO ("no seek needed");
res = TRUE;
{
g_rec_mutex_unlock (&priv->state_lock);
GST_INFO ("pipeline is not seekable");
- return TRUE;
+ return FALSE;
}
not_supported:
{
g_rec_mutex_unlock (&priv->state_lock);
- GST_WARNING ("seek unit %d not supported", range->unit);
+ GST_WARNING ("conversion to npt not supported");
+ return FALSE;
+ }
+preroll_failed:
+ {
+ GST_WARNING ("failed to preroll after seek");
return FALSE;
}
}
static void
-gst_rtsp_media_set_status (GstRTSPMedia * media, GstRTSPMediaStatus status)
+stream_collect_blocking (GstRTSPStream * stream, gboolean * blocked)
{
- GstRTSPMediaPrivate *priv = media->priv;
+ *blocked &= gst_rtsp_stream_is_blocking (stream);
+}
- g_mutex_lock (&priv->lock);
- priv->status = status;
- GST_DEBUG ("setting new status to %d", status);
- g_cond_broadcast (&priv->cond);
- g_mutex_unlock (&priv->lock);
+static gboolean
+media_streams_blocking (GstRTSPMedia * media)
+{
+ gboolean blocking = TRUE;
+
+ g_ptr_array_foreach (media->priv->streams, (GFunc) stream_collect_blocking,
+ &blocking);
+
+ return blocking;
}
-/**
- * gst_rtsp_media_get_status:
- * @media: a #GstRTSPMedia
- *
- * Get the status of @media. When @media is busy preparing, this function waits
- * until @media is prepared or in error.
- *
- * Returns: the status of @media.
- */
-GstRTSPMediaStatus
-gst_rtsp_media_get_status (GstRTSPMedia * media)
+static GstStateChangeReturn
+set_state (GstRTSPMedia * media, GstState state)
{
GstRTSPMediaPrivate *priv = media->priv;
- GstRTSPMediaStatus result;
- gint64 end_time;
+ GstStateChangeReturn ret;
- g_mutex_lock (&priv->lock);
- end_time = g_get_monotonic_time () + 20 * G_TIME_SPAN_SECOND;
- /* while we are preparing, wait */
- while (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING) {
- GST_DEBUG ("waiting for status change");
- if (!g_cond_wait_until (&priv->cond, &priv->lock, end_time)) {
- GST_DEBUG ("timeout, assuming error status");
- priv->status = GST_RTSP_MEDIA_STATUS_ERROR;
- }
- }
- /* could be success or error */
- result = priv->status;
- GST_DEBUG ("got status %d", result);
- g_mutex_unlock (&priv->lock);
+ GST_INFO ("set state to %s for media %p", gst_element_state_get_name (state),
+ media);
+ ret = gst_element_set_state (priv->pipeline, state);
- return result;
+ return ret;
+}
+
+static GstStateChangeReturn
+set_target_state (GstRTSPMedia * media, GstState state, gboolean do_state)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ GstStateChangeReturn ret;
+
+ GST_INFO ("set target state to %s for media %p",
+ gst_element_state_get_name (state), media);
+ priv->target_state = state;
+
+ g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_TARGET_STATE], 0,
+ priv->target_state, NULL);
+
+ if (do_state)
+ ret = set_state (media, state);
+ else
+ ret = GST_STATE_CHANGE_SUCCESS;
+
+ return ret;
}
/* called with state-lock */
switch (type) {
case GST_MESSAGE_STATE_CHANGED:
+ {
+ GstState old, new, pending;
+
+ if (GST_MESSAGE_SRC (message) != GST_OBJECT (priv->pipeline))
+ break;
+
+ gst_message_parse_state_changed (message, &old, &new, &pending);
+
+ GST_DEBUG ("%p: went from %s to %s (pending %s)", media,
+ gst_element_state_get_name (old), gst_element_state_get_name (new),
+ gst_element_state_get_name (pending));
+ if ((priv->transport_mode & GST_RTSP_TRANSPORT_MODE_RECORD)
+ && old == GST_STATE_READY && new == GST_STATE_PAUSED) {
+ GST_INFO ("%p: went to PAUSED, prepared now", media);
+ collect_media_stats (media);
+
+ if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
+ }
+
break;
+ }
case GST_MESSAGE_BUFFERING:
{
gint percent;
/* if the desired state is playing, go back */
if (priv->target_state == GST_STATE_PLAYING) {
GST_INFO ("Buffering done, setting pipeline to PLAYING");
- gst_element_set_state (priv->pipeline, GST_STATE_PLAYING);
+ set_state (media, GST_STATE_PLAYING);
} else {
GST_INFO ("Buffering done");
}
if (priv->target_state == GST_STATE_PLAYING) {
/* we were not buffering but PLAYING, PAUSE the pipeline. */
GST_INFO ("Buffering, setting pipeline to PAUSED ...");
- gst_element_set_state (priv->pipeline, GST_STATE_PAUSED);
+ set_state (media, GST_STATE_PAUSED);
} else {
GST_INFO ("Buffering ...");
}
break;
}
case GST_MESSAGE_ELEMENT:
+ {
+ const GstStructure *s;
+
+ s = gst_message_get_structure (message);
+ if (gst_structure_has_name (s, "GstRTSPStreamBlocking")) {
+ GST_DEBUG ("media received blocking message");
+ if (priv->blocked && media_streams_blocking (media)) {
+ GST_DEBUG ("media is blocking");
+ collect_media_stats (media);
+
+ if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
+ }
+ }
break;
+ }
case GST_MESSAGE_STREAM_STATUS:
break;
case GST_MESSAGE_ASYNC_DONE:
g_object_unref (media);
}
+static GstElement *
+find_payload_element (GstElement * payloader)
+{
+ GstElement *pay = NULL;
+
+ if (GST_IS_BIN (payloader)) {
+ GstIterator *iter;
+ GValue item = { 0 };
+
+ iter = gst_bin_iterate_recurse (GST_BIN (payloader));
+ while (gst_iterator_next (iter, &item) == GST_ITERATOR_OK) {
+ GstElement *element = (GstElement *) g_value_get_object (&item);
+ GstElementClass *eclass = GST_ELEMENT_GET_CLASS (element);
+ const gchar *klass;
+
+ klass =
+ gst_element_class_get_metadata (eclass, GST_ELEMENT_METADATA_KLASS);
+ if (klass == NULL)
+ continue;
+
+ if (strstr (klass, "Payloader") && strstr (klass, "RTP")) {
+ pay = gst_object_ref (element);
+ g_value_unset (&item);
+ break;
+ }
+ g_value_unset (&item);
+ }
+ gst_iterator_free (iter);
+ } else {
+ pay = g_object_ref (payloader);
+ }
+
+ return pay;
+}
+
/* called from streaming threads */
static void
pad_added_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
{
GstRTSPMediaPrivate *priv = media->priv;
GstRTSPStream *stream;
+ GstElement *pay;
- /* FIXME, element is likely not a payloader, find the payloader here */
- stream = gst_rtsp_media_create_stream (media, element, pad);
+ /* find the real payload element */
+ pay = find_payload_element (element);
+ stream = gst_rtsp_media_create_stream (media, pay, pad);
+ gst_object_unref (pay);
GST_INFO ("pad added %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
g_rec_mutex_lock (&priv->state_lock);
+ if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
+ goto not_preparing;
+
+ g_object_set_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream", stream);
+
/* we will be adding elements below that will cause ASYNC_DONE to be
* posted in the bus. We want to ignore those messages until the
* pipeline really prerolled. */
/* join the element in the PAUSED state because this callback is
* called from the streaming thread and it is PAUSED */
- gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
- priv->rtpbin, GST_STATE_PAUSED);
+ if (!gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
+ priv->rtpbin, GST_STATE_PAUSED)) {
+ GST_WARNING ("failed to join bin element");
+ }
priv->adding = FALSE;
g_rec_mutex_unlock (&priv->state_lock);
+
+ return;
+
+ /* ERRORS */
+not_preparing:
+ {
+ gst_rtsp_media_remove_stream (media, stream);
+ g_rec_mutex_unlock (&priv->state_lock);
+ GST_INFO ("ignore pad because we are not preparing");
+ return;
+ }
}
static void
-no_more_pads_cb (GstElement * element, GstRTSPMedia * media)
+pad_removed_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
{
GstRTSPMediaPrivate *priv = media->priv;
+ GstRTSPStream *stream;
+
+ stream = g_object_get_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream");
+ if (stream == NULL)
+ return;
+
+ GST_INFO ("pad removed %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
+
+ g_rec_mutex_lock (&priv->state_lock);
+ gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ gst_rtsp_media_remove_stream (media, stream);
+}
+
+static void
+remove_fakesink (GstRTSPMediaPrivate * priv)
+{
GstElement *fakesink;
g_mutex_lock (&priv->lock);
- GST_INFO ("no more pads");
- if ((fakesink = priv->fakesink)) {
+ if ((fakesink = priv->fakesink))
gst_object_ref (fakesink);
- priv->fakesink = NULL;
- g_mutex_unlock (&priv->lock);
+ priv->fakesink = NULL;
+ g_mutex_unlock (&priv->lock);
+ if (fakesink) {
gst_bin_remove (GST_BIN (priv->pipeline), fakesink);
gst_element_set_state (fakesink, GST_STATE_NULL);
gst_object_unref (fakesink);
}
}
-/**
- * gst_rtsp_media_prepare:
- * @media: a #GstRTSPMedia
- *
- * Prepare @media for streaming. This function will create the objects
- * to manage the streaming. A pipeline must have been set on @media with
- * gst_rtsp_media_take_pipeline().
- *
- * It will preroll the pipeline and collect vital information about the streams
- * such as the duration.
- *
- * Returns: %TRUE on success.
- */
-gboolean
-gst_rtsp_media_prepare (GstRTSPMedia * media)
+static void
+no_more_pads_cb (GstElement * element, GstRTSPMedia * media)
{
- GstRTSPMediaPrivate *priv;
- GstStateChangeReturn ret;
- GstRTSPMediaStatus status;
- guint i;
- GstRTSPMediaClass *klass;
- GstBus *bus;
- GList *walk;
-
- g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
-
- priv = media->priv;
-
- g_rec_mutex_lock (&priv->state_lock);
- priv->prepare_count++;
-
- if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED)
- goto was_prepared;
-
- if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
- goto wait_status;
-
- if (priv->status != GST_RTSP_MEDIA_STATUS_UNPREPARED)
- goto not_unprepared;
-
- if (!priv->reusable && priv->reused)
- goto is_reused;
-
- priv->rtpbin = gst_element_factory_make ("rtpbin", NULL);
- if (priv->rtpbin == NULL)
- goto no_rtpbin;
-
- GST_INFO ("preparing media %p", media);
-
- /* reset some variables */
- priv->is_live = FALSE;
- priv->seekable = FALSE;
- priv->buffering = FALSE;
- /* we're preparing now */
- priv->status = GST_RTSP_MEDIA_STATUS_PREPARING;
-
- bus = gst_pipeline_get_bus (GST_PIPELINE_CAST (priv->pipeline));
-
- /* add the pipeline bus to our custom mainloop */
- priv->source = gst_bus_create_watch (bus);
- gst_object_unref (bus);
-
- g_source_set_callback (priv->source, (GSourceFunc) bus_message,
- g_object_ref (media), (GDestroyNotify) watch_destroyed);
-
- klass = GST_RTSP_MEDIA_GET_CLASS (media);
- priv->id = g_source_attach (priv->source, klass->context);
-
- /* add stuff to the bin */
- gst_bin_add (GST_BIN (priv->pipeline), priv->rtpbin);
-
- /* link streams we already have, other streams might appear when we have
- * dynamic elements */
- for (i = 0; i < priv->streams->len; i++) {
- GstRTSPStream *stream;
-
- stream = g_ptr_array_index (priv->streams, i);
-
- gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
- priv->rtpbin, GST_STATE_NULL);
- }
+ GstRTSPMediaPrivate *priv = media->priv;
- for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
- GstElement *elem = walk->data;
+ GST_INFO ("no more pads");
+ remove_fakesink (priv);
+}
- GST_INFO ("adding callbacks for dynamic element %p", elem);
+typedef struct _DynPaySignalHandlers DynPaySignalHandlers;
- g_signal_connect (elem, "pad-added", (GCallback) pad_added_cb, media);
- g_signal_connect (elem, "no-more-pads", (GCallback) no_more_pads_cb, media);
+struct _DynPaySignalHandlers
+{
+ gulong pad_added_handler;
+ gulong pad_removed_handler;
+ gulong no_more_pads_handler;
+};
- /* we add a fakesink here in order to make the state change async. We remove
- * the fakesink again in the no-more-pads callback. */
- priv->fakesink = gst_element_factory_make ("fakesink", "fakesink");
- gst_bin_add (GST_BIN (priv->pipeline), priv->fakesink);
- }
+static gboolean
+start_preroll (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ GstStateChangeReturn ret;
GST_INFO ("setting pipeline to PAUSED for media %p", media);
/* first go to PAUSED */
- ret = gst_element_set_state (priv->pipeline, GST_STATE_PAUSED);
- priv->target_state = GST_STATE_PAUSED;
+ ret = set_target_state (media, GST_STATE_PAUSED, TRUE);
switch (ret) {
case GST_STATE_CHANGE_SUCCESS:
* seeking query in preroll instead */
priv->seekable = FALSE;
priv->is_live = TRUE;
- ret = gst_element_set_state (priv->pipeline, GST_STATE_PLAYING);
+ if (!(priv->transport_mode & GST_RTSP_TRANSPORT_MODE_RECORD)) {
+ /* start blocked to make sure nothing goes to the sink */
+ media_streams_set_blocked (media, TRUE);
+ }
+ ret = set_state (media, GST_STATE_PLAYING);
if (ret == GST_STATE_CHANGE_FAILURE)
goto state_failed;
break;
case GST_STATE_CHANGE_FAILURE:
goto state_failed;
}
-wait_status:
- g_rec_mutex_unlock (&priv->state_lock);
-
- /* now wait for all pads to be prerolled, FIXME, we should somehow be
- * able to do this async so that we don't block the server thread. */
- status = gst_rtsp_media_get_status (media);
- if (status == GST_RTSP_MEDIA_STATUS_ERROR)
- goto state_failed;
-
- g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_PREPARED], 0, NULL);
-
- GST_INFO ("object %p is prerolled", media);
return TRUE;
- /* OK */
-was_prepared:
- {
- GST_LOG ("media %p was prepared", media);
- g_rec_mutex_unlock (&priv->state_lock);
- return TRUE;
- }
- /* ERRORS */
-not_unprepared:
- {
- GST_WARNING ("media %p was not unprepared", media);
- priv->prepare_count--;
- g_rec_mutex_unlock (&priv->state_lock);
- return FALSE;
- }
-is_reused:
- {
- priv->prepare_count--;
- g_rec_mutex_unlock (&priv->state_lock);
- GST_WARNING ("can not reuse media %p", media);
- return FALSE;
- }
-no_rtpbin:
+state_failed:
{
- priv->prepare_count--;
- g_rec_mutex_unlock (&priv->state_lock);
- GST_WARNING ("no rtpbin element");
- g_warning ("failed to create element 'rtpbin', check your installation");
+ GST_WARNING ("failed to preroll pipeline");
return FALSE;
}
-state_failed:
+}
+
+static gboolean
+wait_preroll (GstRTSPMedia * media)
+{
+ GstRTSPMediaStatus status;
+
+ GST_DEBUG ("wait to preroll pipeline");
+
+ /* wait until pipeline is prerolled */
+ status = gst_rtsp_media_get_status (media);
+ if (status == GST_RTSP_MEDIA_STATUS_ERROR)
+ goto preroll_failed;
+
+ return TRUE;
+
+preroll_failed:
{
GST_WARNING ("failed to preroll pipeline");
- gst_rtsp_media_unprepare (media);
- g_rec_mutex_unlock (&priv->state_lock);
return FALSE;
}
}
-/* must be called with state-lock */
-static void
-finish_unprepare (GstRTSPMedia * media)
+static gboolean
+start_prepare (GstRTSPMedia * media)
{
GstRTSPMediaPrivate *priv = media->priv;
- gint i;
-
- GST_DEBUG ("shutting down");
-
- gst_element_set_state (priv->pipeline, GST_STATE_NULL);
+ guint i;
+ GList *walk;
+ /* link streams we already have, other streams might appear when we have
+ * dynamic elements */
for (i = 0; i < priv->streams->len; i++) {
GstRTSPStream *stream;
- GST_INFO ("Removing elements of stream %d from pipeline", i);
-
stream = g_ptr_array_index (priv->streams, i);
- gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
+ if (!gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
+ priv->rtpbin, GST_STATE_NULL)) {
+ goto join_bin_failed;
+ }
}
- g_ptr_array_set_size (priv->streams, 0);
- gst_bin_remove (GST_BIN (priv->pipeline), priv->rtpbin);
- priv->rtpbin = NULL;
+ if (priv->rtpbin)
+ g_object_set (priv->rtpbin, "do-retransmission", priv->rtx_time > 0, NULL);
- if (priv->nettime)
- gst_object_unref (priv->nettime);
- priv->nettime = NULL;
+ for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
+ GstElement *elem = walk->data;
+ DynPaySignalHandlers *handlers = g_slice_new (DynPaySignalHandlers);
- priv->reused = TRUE;
- priv->status = GST_RTSP_MEDIA_STATUS_UNPREPARED;
+ GST_INFO ("adding callbacks for dynamic element %p", elem);
+
+ handlers->pad_added_handler = g_signal_connect (elem, "pad-added",
+ (GCallback) pad_added_cb, media);
+ handlers->pad_removed_handler = g_signal_connect (elem, "pad-removed",
+ (GCallback) pad_removed_cb, media);
+ handlers->no_more_pads_handler = g_signal_connect (elem, "no-more-pads",
+ (GCallback) no_more_pads_cb, media);
+
+ g_object_set_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers", handlers);
+
+ /* we add a fakesink here in order to make the state change async. We remove
+ * the fakesink again in the no-more-pads callback. */
+ priv->fakesink = gst_element_factory_make ("fakesink", "fakesink");
+ gst_bin_add (GST_BIN (priv->pipeline), priv->fakesink);
+ }
+
+ if (!start_preroll (media))
+ goto preroll_failed;
+
+ return FALSE;
+
+join_bin_failed:
+ {
+ GST_WARNING ("failed to join bin element");
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
+ return FALSE;
+ }
+preroll_failed:
+ {
+ GST_WARNING ("failed to preroll pipeline");
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
+ return FALSE;
+ }
+}
+
+static gboolean
+default_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
+{
+ GstRTSPMediaPrivate *priv;
+ GstRTSPMediaClass *klass;
+ GstBus *bus;
+ GMainContext *context;
+ GSource *source;
+
+ priv = media->priv;
+
+ klass = GST_RTSP_MEDIA_GET_CLASS (media);
+
+ if (!klass->create_rtpbin)
+ goto no_create_rtpbin;
+
+ priv->rtpbin = klass->create_rtpbin (media);
+ if (priv->rtpbin != NULL) {
+ gboolean success = TRUE;
+
+ g_object_set (priv->rtpbin, "latency", priv->latency, NULL);
+
+ if (klass->setup_rtpbin)
+ success = klass->setup_rtpbin (media, priv->rtpbin);
+
+ if (success == FALSE) {
+ gst_object_unref (priv->rtpbin);
+ priv->rtpbin = NULL;
+ }
+ }
+ if (priv->rtpbin == NULL)
+ goto no_rtpbin;
+
+ priv->thread = thread;
+ context = (thread != NULL) ? (thread->context) : NULL;
+
+ bus = gst_pipeline_get_bus (GST_PIPELINE_CAST (priv->pipeline));
+
+ /* add the pipeline bus to our custom mainloop */
+ priv->source = gst_bus_create_watch (bus);
+ gst_object_unref (bus);
+
+ g_source_set_callback (priv->source, (GSourceFunc) bus_message,
+ g_object_ref (media), (GDestroyNotify) watch_destroyed);
+
+ priv->id = g_source_attach (priv->source, context);
+
+ /* add stuff to the bin */
+ gst_bin_add (GST_BIN (priv->pipeline), priv->rtpbin);
+
+ /* do remainder in context */
+ source = g_idle_source_new ();
+ g_source_set_callback (source, (GSourceFunc) start_prepare, media, NULL);
+ g_source_attach (source, context);
+ g_source_unref (source);
+
+ return TRUE;
+
+ /* ERRORS */
+no_create_rtpbin:
+ {
+ GST_ERROR ("no create_rtpbin function");
+ g_critical ("no create_rtpbin vmethod function set");
+ return FALSE;
+ }
+no_rtpbin:
+ {
+ GST_WARNING ("no rtpbin element");
+ g_warning ("failed to create element 'rtpbin', check your installation");
+ return FALSE;
+ }
+}
+
+/**
+ * gst_rtsp_media_prepare:
+ * @media: a #GstRTSPMedia
+ * @thread: (transfer full) (allow-none): a #GstRTSPThread to run the
+ * bus handler or %NULL
+ *
+ * Prepare @media for streaming. This function will create the objects
+ * to manage the streaming. A pipeline must have been set on @media with
+ * gst_rtsp_media_take_pipeline().
+ *
+ * It will preroll the pipeline and collect vital information about the streams
+ * such as the duration.
+ *
+ * Returns: %TRUE on success.
+ */
+gboolean
+gst_rtsp_media_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
+{
+ GstRTSPMediaPrivate *priv;
+ GstRTSPMediaClass *klass;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+ priv = media->priv;
+
+ g_rec_mutex_lock (&priv->state_lock);
+ priv->prepare_count++;
+
+ if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED ||
+ priv->status == GST_RTSP_MEDIA_STATUS_SUSPENDED)
+ goto was_prepared;
+
+ if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
+ goto is_preparing;
+
+ if (priv->status != GST_RTSP_MEDIA_STATUS_UNPREPARED)
+ goto not_unprepared;
+
+ if (!priv->reusable && priv->reused)
+ goto is_reused;
+
+ GST_INFO ("preparing media %p", media);
+
+ /* reset some variables */
+ priv->is_live = FALSE;
+ priv->seekable = FALSE;
+ priv->buffering = FALSE;
+
+ /* we're preparing now */
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
+
+ klass = GST_RTSP_MEDIA_GET_CLASS (media);
+ if (klass->prepare) {
+ if (!klass->prepare (media, thread))
+ goto prepare_failed;
+ }
+
+wait_status:
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ /* now wait for all pads to be prerolled, FIXME, we should somehow be
+ * able to do this async so that we don't block the server thread. */
+ if (!wait_preroll (media))
+ goto preroll_failed;
+
+ g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_PREPARED], 0, NULL);
+
+ GST_INFO ("object %p is prerolled", media);
+
+ return TRUE;
+
+ /* OK */
+is_preparing:
+ {
+ /* we are not going to use the giving thread, so stop it. */
+ if (thread)
+ gst_rtsp_thread_stop (thread);
+ goto wait_status;
+ }
+was_prepared:
+ {
+ GST_LOG ("media %p was prepared", media);
+ /* we are not going to use the giving thread, so stop it. */
+ if (thread)
+ gst_rtsp_thread_stop (thread);
+ g_rec_mutex_unlock (&priv->state_lock);
+ return TRUE;
+ }
+ /* ERRORS */
+not_unprepared:
+ {
+ /* we are not going to use the giving thread, so stop it. */
+ if (thread)
+ gst_rtsp_thread_stop (thread);
+ GST_WARNING ("media %p was not unprepared", media);
+ priv->prepare_count--;
+ g_rec_mutex_unlock (&priv->state_lock);
+ return FALSE;
+ }
+is_reused:
+ {
+ /* we are not going to use the giving thread, so stop it. */
+ if (thread)
+ gst_rtsp_thread_stop (thread);
+ priv->prepare_count--;
+ g_rec_mutex_unlock (&priv->state_lock);
+ GST_WARNING ("can not reuse media %p", media);
+ return FALSE;
+ }
+prepare_failed:
+ {
+ /* we are not going to use the giving thread, so stop it. */
+ if (thread)
+ gst_rtsp_thread_stop (thread);
+ priv->prepare_count--;
+ g_rec_mutex_unlock (&priv->state_lock);
+ GST_ERROR ("failed to prepare media");
+ return FALSE;
+ }
+preroll_failed:
+ {
+ GST_WARNING ("failed to preroll pipeline");
+ gst_rtsp_media_unprepare (media);
+ return FALSE;
+ }
+}
+
+/* must be called with state-lock */
+static void
+finish_unprepare (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ gint i;
+ GList *walk;
+
+ GST_DEBUG ("shutting down");
+
+ /* release the lock on shutdown, otherwise pad_added_cb might try to
+ * acquire the lock and then we deadlock */
+ g_rec_mutex_unlock (&priv->state_lock);
+ set_state (media, GST_STATE_NULL);
+ g_rec_mutex_lock (&priv->state_lock);
+ remove_fakesink (priv);
+
+ for (i = 0; i < priv->streams->len; i++) {
+ GstRTSPStream *stream;
+
+ GST_INFO ("Removing elements of stream %d from pipeline", i);
+
+ stream = g_ptr_array_index (priv->streams, i);
+
+ gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
+ }
+
+ /* remove the pad signal handlers */
+ for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
+ GstElement *elem = walk->data;
+ DynPaySignalHandlers *handlers;
+
+ handlers =
+ g_object_steal_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers");
+ g_assert (handlers != NULL);
+
+ g_signal_handler_disconnect (G_OBJECT (elem), handlers->pad_added_handler);
+ g_signal_handler_disconnect (G_OBJECT (elem),
+ handlers->pad_removed_handler);
+ g_signal_handler_disconnect (G_OBJECT (elem),
+ handlers->no_more_pads_handler);
+
+ g_slice_free (DynPaySignalHandlers, handlers);
+ }
+
+ gst_bin_remove (GST_BIN (priv->pipeline), priv->rtpbin);
+ priv->rtpbin = NULL;
+
+ if (priv->nettime)
+ gst_object_unref (priv->nettime);
+ priv->nettime = NULL;
+
+ priv->reused = TRUE;
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARED);
/* when the media is not reusable, this will effectively unref the media and
* recreate it */
g_source_destroy (priv->source);
g_source_unref (priv->source);
}
+ if (priv->thread) {
+ GST_DEBUG ("stop thread");
+ gst_rtsp_thread_stop (priv->thread);
+ }
}
/* called with state-lock */
{
GstRTSPMediaPrivate *priv = media->priv;
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARING);
+
if (priv->eos_shutdown) {
GST_DEBUG ("sending EOS for shutdown");
/* ref so that we don't disappear */
gst_element_send_event (priv->pipeline, gst_event_new_eos ());
/* we need to go to playing again for the EOS to propagate, normally in this
* state, nothing is receiving data from us anymore so this is ok. */
- gst_element_set_state (priv->pipeline, GST_STATE_PLAYING);
- priv->status = GST_RTSP_MEDIA_STATUS_UNPREPARING;
+ set_state (media, GST_STATE_PLAYING);
} else {
finish_unprepare (media);
}
goto is_busy;
GST_INFO ("unprepare media %p", media);
- priv->target_state = GST_STATE_NULL;
+ if (priv->blocked)
+ media_streams_set_blocked (media, FALSE);
+ set_target_state (media, GST_STATE_NULL, FALSE);
success = TRUE;
if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED) {
*
* @media must be prepared before this method returns a valid clock object.
*
- * Returns: the #GstClock used by @media. unref after usage.
+ * Returns: (transfer full): the #GstClock used by @media. unref after usage.
*/
GstClock *
gst_rtsp_media_get_clock (GstRTSPMedia * media)
/**
* gst_rtsp_media_get_time_provider:
* @media: a #GstRTSPMedia
- * @address: an address or NULL
+ * @address: (allow-none): an address or %NULL
* @port: a port or 0
*
* Get the #GstNetTimeProvider for the clock used by @media. The time provider
* will listen on @address and @port for client time requests.
*
- * Returns: the #GstNetTimeProvider of @media.
+ * Returns: (transfer full): the #GstNetTimeProvider of @media.
*/
GstNetTimeProvider *
gst_rtsp_media_get_time_provider (GstRTSPMedia * media, const gchar * address,
return provider;
}
+static gboolean
+default_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp, GstSDPInfo * info)
+{
+ return gst_rtsp_sdp_from_media (sdp, info, media);
+}
+
/**
- * gst_rtsp_media_set_state:
+ * gst_rtsp_media_setup_sdp:
* @media: a #GstRTSPMedia
- * @state: the target state of the media
- * @transports: a #GPtrArray of #GstRTSPStreamTransport pointers
+ * @sdp: (transfer none): a #GstSDPMessage
+ * @info: (transfer none): a #GstSDPInfo
*
- * Set the state of @media to @state and for the transports in @transports.
- *
- * @media must be prepared with gst_rtsp_media_prepare();
+ * Add @media specific info to @sdp. @info is used to configure the connection
+ * information in the SDP.
*
- * Returns: %TRUE on success.
+ * Returns: TRUE on success.
*/
gboolean
-gst_rtsp_media_set_state (GstRTSPMedia * media, GstState state,
- GPtrArray * transports)
+gst_rtsp_media_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
+ GstSDPInfo * info)
{
GstRTSPMediaPrivate *priv;
- gint i;
- gboolean activate, deactivate, do_state;
- gint old_active;
+ GstRTSPMediaClass *klass;
+ gboolean res;
g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
- g_return_val_if_fail (transports != NULL, FALSE);
+ g_return_val_if_fail (sdp != NULL, FALSE);
+ g_return_val_if_fail (info != NULL, FALSE);
priv = media->priv;
g_rec_mutex_lock (&priv->state_lock);
- if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
- goto not_prepared;
-
- /* NULL and READY are the same */
- if (state == GST_STATE_READY)
- state = GST_STATE_NULL;
- activate = deactivate = FALSE;
+ klass = GST_RTSP_MEDIA_GET_CLASS (media);
- GST_INFO ("going to state %s media %p", gst_element_state_get_name (state),
- media);
+ if (!klass->setup_sdp)
+ goto no_setup_sdp;
- switch (state) {
- case GST_STATE_NULL:
- case GST_STATE_PAUSED:
- /* we're going from PLAYING to PAUSED, READY or NULL, deactivate */
- if (priv->target_state == GST_STATE_PLAYING)
- deactivate = TRUE;
- break;
- case GST_STATE_PLAYING:
- /* we're going to PLAYING, activate */
- activate = TRUE;
- break;
- default:
- break;
- }
- old_active = priv->n_active;
+ res = klass->setup_sdp (media, sdp, info);
- for (i = 0; i < transports->len; i++) {
- GstRTSPStreamTransport *trans;
+ g_rec_mutex_unlock (&priv->state_lock);
- /* we need a non-NULL entry in the array */
- trans = g_ptr_array_index (transports, i);
- if (trans == NULL)
- continue;
+ return res;
+
+ /* ERRORS */
+no_setup_sdp:
+ {
+ g_rec_mutex_unlock (&priv->state_lock);
+ GST_ERROR ("no setup_sdp function");
+ g_critical ("no setup_sdp vmethod function set");
+ return FALSE;
+ }
+}
+
+static const gchar *
+rtsp_get_attribute_for_pt (const GstSDPMedia * media, const gchar * name,
+ gint pt)
+{
+ guint i;
+
+ for (i = 0;; i++) {
+ const gchar *attr;
+ gint val;
+
+ if ((attr = gst_sdp_media_get_attribute_val_n (media, name, i)) == NULL)
+ break;
+
+ if (sscanf (attr, "%d ", &val) != 1)
+ continue;
+
+ if (val == pt)
+ return attr;
+ }
+ return NULL;
+}
+
+#define PARSE_INT(p, del, res) \
+G_STMT_START { \
+ gchar *t = p; \
+ p = strstr (p, del); \
+ if (p == NULL) \
+ res = -1; \
+ else { \
+ *p = '\0'; \
+ p++; \
+ res = atoi (t); \
+ } \
+} G_STMT_END
+
+#define PARSE_STRING(p, del, res) \
+G_STMT_START { \
+ gchar *t = p; \
+ p = strstr (p, del); \
+ if (p == NULL) { \
+ res = NULL; \
+ p = t; \
+ } \
+ else { \
+ *p = '\0'; \
+ p++; \
+ res = t; \
+ } \
+} G_STMT_END
+
+#define SKIP_SPACES(p) \
+ while (*p && g_ascii_isspace (*p)) \
+ p++;
+
+/* rtpmap contains:
+ *
+ * <payload> <encoding_name>/<clock_rate>[/<encoding_params>]
+ */
+static gboolean
+parse_rtpmap (const gchar * rtpmap, gint * payload, gchar ** name,
+ gint * rate, gchar ** params)
+{
+ gchar *p, *t;
+
+ p = (gchar *) rtpmap;
+
+ PARSE_INT (p, " ", *payload);
+ if (*payload == -1)
+ return FALSE;
+
+ SKIP_SPACES (p);
+ if (*p == '\0')
+ return FALSE;
+
+ PARSE_STRING (p, "/", *name);
+ if (*name == NULL) {
+ GST_DEBUG ("no rate, name %s", p);
+ /* no rate, assume -1 then, this is not supposed to happen but RealMedia
+ * streams seem to omit the rate. */
+ *name = p;
+ *rate = -1;
+ return TRUE;
+ }
+
+ t = p;
+ p = strstr (p, "/");
+ if (p == NULL) {
+ *rate = atoi (t);
+ return TRUE;
+ }
+ *p = '\0';
+ p++;
+ *rate = atoi (t);
+
+ t = p;
+ if (*p == '\0')
+ return TRUE;
+ *params = t;
+
+ return TRUE;
+}
+
+/*
+ * Mapping of caps to and from SDP fields:
+ *
+ * a=rtpmap:<payload> <encoding_name>/<clock_rate>[/<encoding_params>]
+ * a=fmtp:<payload> <param>[=<value>];...
+ */
+static GstCaps *
+media_to_caps (gint pt, const GstSDPMedia * media)
+{
+ GstCaps *caps;
+ const gchar *rtpmap;
+ const gchar *fmtp;
+ gchar *name = NULL;
+ gint rate = -1;
+ gchar *params = NULL;
+ gchar *tmp;
+ GstStructure *s;
+ gint payload = 0;
+ gboolean ret;
+
+ /* get and parse rtpmap */
+ rtpmap = rtsp_get_attribute_for_pt (media, "rtpmap", pt);
+
+ if (rtpmap) {
+ ret = parse_rtpmap (rtpmap, &payload, &name, &rate, ¶ms);
+ if (!ret) {
+ g_warning ("error parsing rtpmap, ignoring");
+ rtpmap = NULL;
+ }
+ }
+ /* dynamic payloads need rtpmap or we fail */
+ if (rtpmap == NULL && pt >= 96)
+ goto no_rtpmap;
+
+ /* check if we have a rate, if not, we need to look up the rate from the
+ * default rates based on the payload types. */
+ if (rate == -1) {
+ const GstRTPPayloadInfo *info;
+
+ if (GST_RTP_PAYLOAD_IS_DYNAMIC (pt)) {
+ /* dynamic types, use media and encoding_name */
+ tmp = g_ascii_strdown (media->media, -1);
+ info = gst_rtp_payload_info_for_name (tmp, name);
+ g_free (tmp);
+ } else {
+ /* static types, use payload type */
+ info = gst_rtp_payload_info_for_pt (pt);
+ }
+
+ if (info) {
+ if ((rate = info->clock_rate) == 0)
+ rate = -1;
+ }
+ /* we fail if we cannot find one */
+ if (rate == -1)
+ goto no_rate;
+ }
+
+ tmp = g_ascii_strdown (media->media, -1);
+ caps = gst_caps_new_simple ("application/x-unknown",
+ "media", G_TYPE_STRING, tmp, "payload", G_TYPE_INT, pt, NULL);
+ g_free (tmp);
+ s = gst_caps_get_structure (caps, 0);
+
+ gst_structure_set (s, "clock-rate", G_TYPE_INT, rate, NULL);
+
+ /* encoding name must be upper case */
+ if (name != NULL) {
+ tmp = g_ascii_strup (name, -1);
+ gst_structure_set (s, "encoding-name", G_TYPE_STRING, tmp, NULL);
+ g_free (tmp);
+ }
+
+ /* params must be lower case */
+ if (params != NULL) {
+ tmp = g_ascii_strdown (params, -1);
+ gst_structure_set (s, "encoding-params", G_TYPE_STRING, tmp, NULL);
+ g_free (tmp);
+ }
+
+ /* parse optional fmtp: field */
+ if ((fmtp = rtsp_get_attribute_for_pt (media, "fmtp", pt))) {
+ gchar *p;
+ gint payload = 0;
+
+ p = (gchar *) fmtp;
+
+ /* p is now of the format <payload> <param>[=<value>];... */
+ PARSE_INT (p, " ", payload);
+ if (payload != -1 && payload == pt) {
+ gchar **pairs;
+ gint i;
+
+ /* <param>[=<value>] are separated with ';' */
+ pairs = g_strsplit (p, ";", 0);
+ for (i = 0; pairs[i]; i++) {
+ gchar *valpos;
+ const gchar *val, *key;
+
+ /* the key may not have a '=', the value can have other '='s */
+ valpos = strstr (pairs[i], "=");
+ if (valpos) {
+ /* we have a '=' and thus a value, remove the '=' with \0 */
+ *valpos = '\0';
+ /* value is everything between '=' and ';'. We split the pairs at ;
+ * boundaries so we can take the remainder of the value. Some servers
+ * put spaces around the value which we strip off here. Alternatively
+ * we could strip those spaces in the depayloaders should these spaces
+ * actually carry any meaning in the future. */
+ val = g_strstrip (valpos + 1);
+ } else {
+ /* simple <param>;.. is translated into <param>=1;... */
+ val = "1";
+ }
+ /* strip the key of spaces, convert key to lowercase but not the value. */
+ key = g_strstrip (pairs[i]);
+ if (strlen (key) > 1) {
+ tmp = g_ascii_strdown (key, -1);
+ gst_structure_set (s, tmp, G_TYPE_STRING, val, NULL);
+ g_free (tmp);
+ }
+ }
+ g_strfreev (pairs);
+ }
+ }
+ return caps;
+
+ /* ERRORS */
+no_rtpmap:
+ {
+ g_warning ("rtpmap type not given for dynamic payload %d", pt);
+ return NULL;
+ }
+no_rate:
+ {
+ g_warning ("rate unknown for payload type %d", pt);
+ return NULL;
+ }
+}
+
+static gboolean
+parse_keymgmt (const gchar * keymgmt, GstCaps * caps)
+{
+ gboolean res = FALSE;
+ gchar *p, *kmpid;
+ gsize size;
+ guchar *data;
+ GstMIKEYMessage *msg;
+ const GstMIKEYPayload *payload;
+ const gchar *srtp_cipher;
+ const gchar *srtp_auth;
+
+ p = (gchar *) keymgmt;
+
+ SKIP_SPACES (p);
+ if (*p == '\0')
+ return FALSE;
+
+ PARSE_STRING (p, " ", kmpid);
+ if (!g_str_equal (kmpid, "mikey"))
+ return FALSE;
+
+ data = g_base64_decode (p, &size);
+ if (data == NULL)
+ return FALSE;
+
+ msg = gst_mikey_message_new_from_data (data, size, NULL, NULL);
+ g_free (data);
+ if (msg == NULL)
+ return FALSE;
+
+ srtp_cipher = "aes-128-icm";
+ srtp_auth = "hmac-sha1-80";
+
+ /* check the Security policy if any */
+ if ((payload = gst_mikey_message_find_payload (msg, GST_MIKEY_PT_SP, 0))) {
+ GstMIKEYPayloadSP *p = (GstMIKEYPayloadSP *) payload;
+ guint len, i;
+
+ if (p->proto != GST_MIKEY_SEC_PROTO_SRTP)
+ goto done;
+
+ len = gst_mikey_payload_sp_get_n_params (payload);
+ for (i = 0; i < len; i++) {
+ const GstMIKEYPayloadSPParam *param =
+ gst_mikey_payload_sp_get_param (payload, i);
+
+ switch (param->type) {
+ case GST_MIKEY_SP_SRTP_ENC_ALG:
+ switch (param->val[0]) {
+ case 0:
+ srtp_cipher = "null";
+ break;
+ case 2:
+ case 1:
+ srtp_cipher = "aes-128-icm";
+ break;
+ default:
+ break;
+ }
+ break;
+ case GST_MIKEY_SP_SRTP_ENC_KEY_LEN:
+ switch (param->val[0]) {
+ case AES_128_KEY_LEN:
+ srtp_cipher = "aes-128-icm";
+ break;
+ case AES_256_KEY_LEN:
+ srtp_cipher = "aes-256-icm";
+ break;
+ default:
+ break;
+ }
+ break;
+ case GST_MIKEY_SP_SRTP_AUTH_ALG:
+ switch (param->val[0]) {
+ case 0:
+ srtp_auth = "null";
+ break;
+ case 2:
+ case 1:
+ srtp_auth = "hmac-sha1-80";
+ break;
+ default:
+ break;
+ }
+ break;
+ case GST_MIKEY_SP_SRTP_AUTH_KEY_LEN:
+ switch (param->val[0]) {
+ case HMAC_32_KEY_LEN:
+ srtp_auth = "hmac-sha1-32";
+ break;
+ case HMAC_80_KEY_LEN:
+ srtp_auth = "hmac-sha1-80";
+ break;
+ default:
+ break;
+ }
+ break;
+ case GST_MIKEY_SP_SRTP_SRTP_ENC:
+ break;
+ case GST_MIKEY_SP_SRTP_SRTCP_ENC:
+ break;
+ default:
+ break;
+ }
+ }
+ }
+
+ if (!(payload = gst_mikey_message_find_payload (msg, GST_MIKEY_PT_KEMAC, 0)))
+ goto done;
+ else {
+ GstMIKEYPayloadKEMAC *p = (GstMIKEYPayloadKEMAC *) payload;
+ const GstMIKEYPayload *sub;
+ GstMIKEYPayloadKeyData *pkd;
+ GstBuffer *buf;
+
+ if (p->enc_alg != GST_MIKEY_ENC_NULL || p->mac_alg != GST_MIKEY_MAC_NULL)
+ goto done;
+
+ if (!(sub = gst_mikey_payload_kemac_get_sub (payload, 0)))
+ goto done;
+
+ if (sub->type != GST_MIKEY_PT_KEY_DATA)
+ goto done;
+
+ pkd = (GstMIKEYPayloadKeyData *) sub;
+ buf =
+ gst_buffer_new_wrapped (g_memdup (pkd->key_data, pkd->key_len),
+ pkd->key_len);
+ gst_caps_set_simple (caps, "srtp-key", GST_TYPE_BUFFER, buf, NULL);
+ }
+
+ gst_caps_set_simple (caps,
+ "srtp-cipher", G_TYPE_STRING, srtp_cipher,
+ "srtp-auth", G_TYPE_STRING, srtp_auth,
+ "srtcp-cipher", G_TYPE_STRING, srtp_cipher,
+ "srtcp-auth", G_TYPE_STRING, srtp_auth, NULL);
+
+ res = TRUE;
+done:
+ gst_mikey_message_unref (msg);
+
+ return res;
+}
+
+/*
+ * Mapping SDP attributes to caps
+ *
+ * prepend 'a-' to IANA registered sdp attributes names
+ * (ie: not prefixed with 'x-') in order to avoid
+ * collision with gstreamer standard caps properties names
+ */
+static void
+sdp_attributes_to_caps (GArray * attributes, GstCaps * caps)
+{
+ if (attributes->len > 0) {
+ GstStructure *s;
+ guint i;
+
+ s = gst_caps_get_structure (caps, 0);
+
+ for (i = 0; i < attributes->len; i++) {
+ GstSDPAttribute *attr = &g_array_index (attributes, GstSDPAttribute, i);
+ gchar *tofree, *key;
+
+ key = attr->key;
+
+ /* skip some of the attribute we already handle */
+ if (!strcmp (key, "fmtp"))
+ continue;
+ if (!strcmp (key, "rtpmap"))
+ continue;
+ if (!strcmp (key, "control"))
+ continue;
+ if (!strcmp (key, "range"))
+ continue;
+ if (g_str_equal (key, "key-mgmt")) {
+ parse_keymgmt (attr->value, caps);
+ continue;
+ }
+
+ /* string must be valid UTF8 */
+ if (!g_utf8_validate (attr->value, -1, NULL))
+ continue;
+
+ if (!g_str_has_prefix (key, "x-"))
+ tofree = key = g_strdup_printf ("a-%s", key);
+ else
+ tofree = NULL;
+
+ GST_DEBUG ("adding caps: %s=%s", key, attr->value);
+ gst_structure_set (s, key, G_TYPE_STRING, attr->value, NULL);
+ g_free (tofree);
+ }
+ }
+}
+
+static gboolean
+default_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ gint i, medias_len;
+
+ medias_len = gst_sdp_message_medias_len (sdp);
+ if (medias_len != priv->streams->len) {
+ GST_ERROR ("%p: Media has more or less streams than SDP (%d /= %d)", media,
+ priv->streams->len, medias_len);
+ return FALSE;
+ }
+
+ for (i = 0; i < medias_len; i++) {
+ const gchar *proto, *media_type;
+ const GstSDPMedia *sdp_media = gst_sdp_message_get_media (sdp, i);
+ GstRTSPStream *stream;
+ gint j, formats_len;
+ const gchar *control;
+ GstRTSPProfile profile, profiles;
+
+ stream = g_ptr_array_index (priv->streams, i);
+
+ /* TODO: Should we do something with the other SDP information? */
+
+ /* get proto */
+ proto = gst_sdp_media_get_proto (sdp_media);
+ if (proto == NULL) {
+ GST_ERROR ("%p: SDP media %d has no proto", media, i);
+ return FALSE;
+ }
+
+ if (g_str_equal (proto, "RTP/AVP")) {
+ media_type = "application/x-rtp";
+ profile = GST_RTSP_PROFILE_AVP;
+ } else if (g_str_equal (proto, "RTP/SAVP")) {
+ media_type = "application/x-srtp";
+ profile = GST_RTSP_PROFILE_SAVP;
+ } else if (g_str_equal (proto, "RTP/AVPF")) {
+ media_type = "application/x-rtp";
+ profile = GST_RTSP_PROFILE_AVPF;
+ } else if (g_str_equal (proto, "RTP/SAVPF")) {
+ media_type = "application/x-srtp";
+ profile = GST_RTSP_PROFILE_SAVPF;
+ } else {
+ GST_ERROR ("%p: unsupported profile '%s' for stream %d", media, proto, i);
+ return FALSE;
+ }
+
+ profiles = gst_rtsp_stream_get_profiles (stream);
+ if ((profiles & profile) == 0) {
+ GST_ERROR ("%p: unsupported profile '%s' for stream %d", media, proto, i);
+ return FALSE;
+ }
+
+ formats_len = gst_sdp_media_formats_len (sdp_media);
+ for (j = 0; j < formats_len; j++) {
+ gint pt;
+ GstCaps *caps;
+ GstStructure *s;
+
+ pt = atoi (gst_sdp_media_get_format (sdp_media, j));
+
+ GST_DEBUG (" looking at %d pt: %d", j, pt);
+
+ /* convert caps */
+ caps = media_to_caps (pt, sdp_media);
+ if (caps == NULL) {
+ GST_WARNING (" skipping pt %d without caps", pt);
+ continue;
+ }
+
+ /* do some tweaks */
+ GST_DEBUG ("mapping sdp session level attributes to caps");
+ sdp_attributes_to_caps (sdp->attributes, caps);
+ GST_DEBUG ("mapping sdp media level attributes to caps");
+ sdp_attributes_to_caps (sdp_media->attributes, caps);
+
+ s = gst_caps_get_structure (caps, 0);
+ gst_structure_set_name (s, media_type);
+
+ gst_rtsp_stream_set_pt_map (stream, pt, caps);
+ gst_caps_unref (caps);
+ }
+
+ control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+ if (control)
+ gst_rtsp_stream_set_control (stream, control);
+
+ }
+
+ return TRUE;
+}
+
+/**
+ * gst_rtsp_media_handle_sdp:
+ * @media: a #GstRTSPMedia
+ * @sdp: (transfer none): a #GstSDPMessage
+ *
+ * Configure an SDP on @media for receiving streams
+ *
+ * Returns: TRUE on success.
+ */
+gboolean
+gst_rtsp_media_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp)
+{
+ GstRTSPMediaPrivate *priv;
+ GstRTSPMediaClass *klass;
+ gboolean res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+ g_return_val_if_fail (sdp != NULL, FALSE);
+
+ priv = media->priv;
+
+ g_rec_mutex_lock (&priv->state_lock);
+
+ klass = GST_RTSP_MEDIA_GET_CLASS (media);
+
+ if (!klass->handle_sdp)
+ goto no_handle_sdp;
+
+ res = klass->handle_sdp (media, sdp);
+
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ return res;
+
+ /* ERRORS */
+no_handle_sdp:
+ {
+ g_rec_mutex_unlock (&priv->state_lock);
+ GST_ERROR ("no handle_sdp function");
+ g_critical ("no handle_sdp vmethod function set");
+ return FALSE;
+ }
+}
+
+static void
+do_set_seqnum (GstRTSPStream * stream)
+{
+ guint16 seq_num;
+ seq_num = gst_rtsp_stream_get_current_seqnum (stream);
+ gst_rtsp_stream_set_seqnum_offset (stream, seq_num + 1);
+}
+
+/* call with state_lock */
+gboolean
+default_suspend (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ GstStateChangeReturn ret;
+
+ switch (priv->suspend_mode) {
+ case GST_RTSP_SUSPEND_MODE_NONE:
+ GST_DEBUG ("media %p no suspend", media);
+ break;
+ case GST_RTSP_SUSPEND_MODE_PAUSE:
+ GST_DEBUG ("media %p suspend to PAUSED", media);
+ ret = set_target_state (media, GST_STATE_PAUSED, TRUE);
+ if (ret == GST_STATE_CHANGE_FAILURE)
+ goto state_failed;
+ break;
+ case GST_RTSP_SUSPEND_MODE_RESET:
+ GST_DEBUG ("media %p suspend to NULL", media);
+ ret = set_target_state (media, GST_STATE_NULL, TRUE);
+ if (ret == GST_STATE_CHANGE_FAILURE)
+ goto state_failed;
+ /* Because payloader needs to set the sequence number as
+ * monotonic, we need to preserve the sequence number
+ * after pause. (otherwise going from pause to play, which
+ * is actually from NULL to PLAY will create a new sequence
+ * number. */
+ g_ptr_array_foreach (priv->streams, (GFunc) do_set_seqnum, NULL);
+ break;
+ default:
+ break;
+ }
+
+ /* let the streams do the state changes freely, if any */
+ media_streams_set_blocked (media, FALSE);
+
+ return TRUE;
+
+ /* ERRORS */
+state_failed:
+ {
+ GST_WARNING ("failed changing pipeline's state for media %p", media);
+ return FALSE;
+ }
+}
+
+/**
+ * gst_rtsp_media_suspend:
+ * @media: a #GstRTSPMedia
+ *
+ * Suspend @media. The state of the pipeline managed by @media is set to
+ * GST_STATE_NULL but all streams are kept. @media can be prepared again
+ * with gst_rtsp_media_unsuspend()
+ *
+ * @media must be prepared with gst_rtsp_media_prepare();
+ *
+ * Returns: %TRUE on success.
+ */
+gboolean
+gst_rtsp_media_suspend (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ GstRTSPMediaClass *klass;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+ GST_FIXME ("suspend for dynamic pipelines needs fixing");
+
+ g_rec_mutex_lock (&priv->state_lock);
+ if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
+ goto not_prepared;
+
+ /* don't attempt to suspend when something is busy */
+ if (priv->n_active > 0)
+ goto done;
+
+ klass = GST_RTSP_MEDIA_GET_CLASS (media);
+ if (klass->suspend) {
+ if (!klass->suspend (media))
+ goto suspend_failed;
+ }
+
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_SUSPENDED);
+done:
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ return TRUE;
+
+ /* ERRORS */
+not_prepared:
+ {
+ g_rec_mutex_unlock (&priv->state_lock);
+ GST_WARNING ("media %p was not prepared", media);
+ return FALSE;
+ }
+suspend_failed:
+ {
+ g_rec_mutex_unlock (&priv->state_lock);
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
+ GST_WARNING ("failed to suspend media %p", media);
+ return FALSE;
+ }
+}
+
+/* call with state_lock */
+gboolean
+default_unsuspend (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+
+ switch (priv->suspend_mode) {
+ case GST_RTSP_SUSPEND_MODE_NONE:
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
+ break;
+ case GST_RTSP_SUSPEND_MODE_PAUSE:
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
+ break;
+ case GST_RTSP_SUSPEND_MODE_RESET:
+ {
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
+ if (!start_preroll (media))
+ goto start_failed;
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ if (!wait_preroll (media))
+ goto preroll_failed;
+
+ g_rec_mutex_lock (&priv->state_lock);
+ }
+ default:
+ break;
+ }
+
+ return TRUE;
+
+ /* ERRORS */
+start_failed:
+ {
+ GST_WARNING ("failed to preroll pipeline");
+ return FALSE;
+ }
+preroll_failed:
+ {
+ GST_WARNING ("failed to preroll pipeline");
+ return FALSE;
+ }
+}
+
+/**
+ * gst_rtsp_media_unsuspend:
+ * @media: a #GstRTSPMedia
+ *
+ * Unsuspend @media if it was in a suspended state. This method does nothing
+ * when the media was not in the suspended state.
+ *
+ * Returns: %TRUE on success.
+ */
+gboolean
+gst_rtsp_media_unsuspend (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ GstRTSPMediaClass *klass;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+ g_rec_mutex_lock (&priv->state_lock);
+ if (priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
+ goto done;
+
+ klass = GST_RTSP_MEDIA_GET_CLASS (media);
+ if (klass->unsuspend) {
+ if (!klass->unsuspend (media))
+ goto unsuspend_failed;
+ }
+
+done:
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ return TRUE;
+
+ /* ERRORS */
+unsuspend_failed:
+ {
+ g_rec_mutex_unlock (&priv->state_lock);
+ GST_WARNING ("failed to unsuspend media %p", media);
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
+ return FALSE;
+ }
+}
+
+/* must be called with state-lock */
+static void
+media_set_pipeline_state_locked (GstRTSPMedia * media, GstState state)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+
+ if (state == GST_STATE_NULL) {
+ gst_rtsp_media_unprepare (media);
+ } else {
+ GST_INFO ("state %s media %p", gst_element_state_get_name (state), media);
+ set_target_state (media, state, FALSE);
+ /* when we are buffering, don't update the state yet, this will be done
+ * when buffering finishes */
+ if (priv->buffering) {
+ GST_INFO ("Buffering busy, delay state change");
+ } else {
+ if (state == GST_STATE_PLAYING)
+ /* make sure pads are not blocking anymore when going to PLAYING */
+ media_streams_set_blocked (media, FALSE);
+
+ set_state (media, state);
+
+ /* and suspend after pause */
+ if (state == GST_STATE_PAUSED)
+ gst_rtsp_media_suspend (media);
+ }
+ }
+}
+
+/**
+ * gst_rtsp_media_set_pipeline_state:
+ * @media: a #GstRTSPMedia
+ * @state: the target state of the pipeline
+ *
+ * Set the state of the pipeline managed by @media to @state
+ */
+void
+gst_rtsp_media_set_pipeline_state (GstRTSPMedia * media, GstState state)
+{
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+ g_rec_mutex_lock (&media->priv->state_lock);
+ media_set_pipeline_state_locked (media, state);
+ g_rec_mutex_unlock (&media->priv->state_lock);
+}
+
+/**
+ * gst_rtsp_media_set_state:
+ * @media: a #GstRTSPMedia
+ * @state: the target state of the media
+ * @transports: (transfer none) (element-type GstRtspServer.RTSPStreamTransport):
+ * a #GPtrArray of #GstRTSPStreamTransport pointers
+ *
+ * Set the state of @media to @state and for the transports in @transports.
+ *
+ * @media must be prepared with gst_rtsp_media_prepare();
+ *
+ * Returns: %TRUE on success.
+ */
+gboolean
+gst_rtsp_media_set_state (GstRTSPMedia * media, GstState state,
+ GPtrArray * transports)
+{
+ GstRTSPMediaPrivate *priv;
+ gint i;
+ gboolean activate, deactivate, do_state;
+ gint old_active;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+ g_return_val_if_fail (transports != NULL, FALSE);
+
+ priv = media->priv;
+
+ g_rec_mutex_lock (&priv->state_lock);
+ if (priv->status == GST_RTSP_MEDIA_STATUS_ERROR)
+ goto error_status;
+ if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
+ priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
+ goto not_prepared;
+
+ /* NULL and READY are the same */
+ if (state == GST_STATE_READY)
+ state = GST_STATE_NULL;
+
+ activate = deactivate = FALSE;
+
+ GST_INFO ("going to state %s media %p, target state %s",
+ gst_element_state_get_name (state), media,
+ gst_element_state_get_name (priv->target_state));
+
+ switch (state) {
+ case GST_STATE_NULL:
+ /* we're going from PLAYING or PAUSED to READY or NULL, deactivate */
+ if (priv->target_state >= GST_STATE_PAUSED)
+ deactivate = TRUE;
+ break;
+ case GST_STATE_PAUSED:
+ /* we're going from PLAYING to PAUSED, READY or NULL, deactivate */
+ if (priv->target_state == GST_STATE_PLAYING)
+ deactivate = TRUE;
+ break;
+ case GST_STATE_PLAYING:
+ /* we're going to PLAYING, activate */
+ activate = TRUE;
+ break;
+ default:
+ break;
+ }
+ old_active = priv->n_active;
+
+ GST_DEBUG ("%d transports, activate %d, deactivate %d", transports->len,
+ activate, deactivate);
+ for (i = 0; i < transports->len; i++) {
+ GstRTSPStreamTransport *trans;
+
+ /* we need a non-NULL entry in the array */
+ trans = g_ptr_array_index (transports, i);
+ if (trans == NULL)
+ continue;
if (activate) {
if (gst_rtsp_stream_transport_set_active (trans, TRUE))
media, do_state);
if (priv->target_state != state) {
- if (do_state) {
- if (state == GST_STATE_NULL) {
- gst_rtsp_media_unprepare (media);
- } else {
- GST_INFO ("state %s media %p", gst_element_state_get_name (state),
- media);
- priv->target_state = state;
- /* when we are buffering, don't update the state yet, this will be done
- * when buffering finishes */
- if (priv->buffering) {
- GST_INFO ("Buffering busy, delay state change");
- } else {
- gst_element_set_state (priv->pipeline, state);
- }
- }
- }
+ if (do_state)
+ media_set_pipeline_state_locked (media, state);
+
g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STATE], 0, state,
NULL);
}
g_rec_mutex_unlock (&priv->state_lock);
return FALSE;
}
+error_status:
+ {
+ GST_WARNING ("media %p in error status while changing to state %d",
+ media, state);
+ if (state == GST_STATE_NULL) {
+ for (i = 0; i < transports->len; i++) {
+ GstRTSPStreamTransport *trans;
+
+ /* we need a non-NULL entry in the array */
+ trans = g_ptr_array_index (transports, i);
+ if (trans == NULL)
+ continue;
+
+ gst_rtsp_stream_transport_set_active (trans, FALSE);
+ }
+ priv->n_active = 0;
+ }
+ g_rec_mutex_unlock (&priv->state_lock);
+ return FALSE;
+ }
+}
+
+/**
+ * gst_rtsp_media_set_transport_mode:
+ * @media: a #GstRTSPMedia
+ * @mode: the new value
+ *
+ * Sets if the media pipeline can work in PLAY or RECORD mode
+ */
+void
+gst_rtsp_media_set_transport_mode (GstRTSPMedia * media,
+ GstRTSPTransportMode mode)
+{
+ GstRTSPMediaPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->transport_mode = mode;
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_get_transport_mode:
+ * @media: a #GstRTSPMedia
+ *
+ * Check if the pipeline for @media can be used for PLAY or RECORD methods.
+ *
+ * Returns: The transport mode.
+ */
+GstRTSPTransportMode
+gst_rtsp_media_get_transport_mode (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ GstRTSPTransportMode res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ res = priv->transport_mode;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
}
projects / platform / upstream / gstreamer.git / blobdiff