#define GST_RTSP_STREAM_GET_PRIVATE(obj) \
(G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_STREAM, GstRTSPStreamPrivate))
+typedef struct
+{
+ GstRTSPStreamTransport *transport;
+
+ /* RTP and RTCP source */
+ GstElement *udpsrc[2];
+ GstPad *selpad[2];
+} GstRTSPMulticastTransportSource;
+
struct _GstRTSPStreamPrivate
{
GMutex lock;
gboolean is_joined;
gchar *control;
+ GstRTSPProfile profiles;
GstRTSPLowerTrans protocols;
/* pads on the rtpbin */
/* the RTPSession object */
GObject *session;
+ /* SRTP encoder/decoder */
+ GstElement *srtpenc;
+ GstElement *srtpdec;
+ GHashTable *keys;
+
/* sinks used for sending and receiving RTP and RTCP over ipv4, they share
* sockets */
GstElement *udpsrc_v4[2];
/* transports we stream to */
guint n_active;
GList *transports;
+ guint transports_cookie;
+ GList *tr_cache;
+ guint tr_cache_cookie;
+
+ /* UDP sources for UDP multicast transports */
+ GList *transport_sources;
gint dscp_qos;
+
+ /* stream blocking */
+ gulong blocked_id;
+ gboolean blocking;
};
#define DEFAULT_CONTROL NULL
+#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
{
PROP_0,
PROP_CONTROL,
+ PROP_PROFILES,
PROP_PROTOCOLS,
PROP_LAST
};
+enum
+{
+ SIGNAL_NEW_RTP_ENCODER,
+ SIGNAL_NEW_RTCP_ENCODER,
+ SIGNAL_LAST
+};
+
GST_DEBUG_CATEGORY_STATIC (rtsp_stream_debug);
#define GST_CAT_DEFAULT rtsp_stream_debug
static void gst_rtsp_stream_finalize (GObject * obj);
+static guint gst_rtsp_stream_signals[SIGNAL_LAST] = { 0 };
+
G_DEFINE_TYPE (GstRTSPStream, gst_rtsp_stream, G_TYPE_OBJECT);
static void
"The control string for this stream", DEFAULT_CONTROL,
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,
DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ gst_rtsp_stream_signals[SIGNAL_NEW_RTP_ENCODER] =
+ g_signal_new ("new-rtp-encoder", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, 0, NULL, NULL, g_cclosure_marshal_generic,
+ G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
+
+ gst_rtsp_stream_signals[SIGNAL_NEW_RTCP_ENCODER] =
+ g_signal_new ("new-rtcp-encoder", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, 0, NULL, NULL, g_cclosure_marshal_generic,
+ G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
+
GST_DEBUG_CATEGORY_INIT (rtsp_stream_debug, "rtspstream", 0, "GstRTSPStream");
ssrc_stream_map_key = g_quark_from_static_string ("GstRTSPServer.stream");
priv->dscp_qos = -1;
priv->control = g_strdup (DEFAULT_CONTROL);
+ priv->profiles = DEFAULT_PROFILES;
priv->protocols = DEFAULT_PROTOCOLS;
g_mutex_init (&priv->lock);
+
+ priv->keys = g_hash_table_new_full (g_direct_hash, g_direct_equal,
+ NULL, (GDestroyNotify) gst_caps_unref);
}
static void
g_free (priv->control);
g_mutex_clear (&priv->lock);
+ g_hash_table_unref (priv->keys);
+
G_OBJECT_CLASS (gst_rtsp_stream_parent_class)->finalize (obj);
}
case PROP_CONTROL:
g_value_take_string (value, gst_rtsp_stream_get_control (stream));
break;
+ case PROP_PROFILES:
+ g_value_set_flags (value, gst_rtsp_stream_get_profiles (stream));
+ break;
case PROP_PROTOCOLS:
g_value_set_flags (value, gst_rtsp_stream_get_protocols (stream));
break;
case PROP_CONTROL:
gst_rtsp_stream_set_control (stream, g_value_get_string (value));
break;
+ case PROP_PROFILES:
+ gst_rtsp_stream_set_profiles (stream, g_value_get_flags (value));
+ break;
case PROP_PROTOCOLS:
gst_rtsp_stream_set_protocols (stream, g_value_get_flags (value));
break;
* Create a new media stream with index @idx that handles RTP data on
* @srcpad and has a payloader element @payloader.
*
- * Returns: a new #GstRTSPStream
+ * Returns: (transfer full): a new #GstRTSPStream
*/
GstRTSPStream *
gst_rtsp_stream_new (guint idx, GstElement * payloader, GstPad * srcpad)
}
/**
+ * gst_rtsp_stream_get_pt:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the stream payload type.
+ *
+ * Return: the stream payload type.
+ */
+guint
+gst_rtsp_stream_get_pt (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+ guint pt;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
+
+ priv = stream->priv;
+
+ g_object_get (G_OBJECT (priv->payloader), "pt", &pt, NULL);
+
+ return pt;
+}
+
+/**
* gst_rtsp_stream_get_srcpad:
* @stream: a #GstRTSPStream
*
* Get the srcpad associated with @stream.
*
- * Return: the srcpad. Unref after usage.
+ * Returns: (transfer full): the srcpad. Unref after usage.
*/
GstPad *
gst_rtsp_stream_get_srcpad (GstRTSPStream * stream)
*
* Get the control string to identify this stream.
*
- * Return: the control string. free after usage.
+ * Returns: (transfer full): the control string. g_free() after usage.
*/
gchar *
gst_rtsp_stream_get_control (GstRTSPStream * stream)
g_mutex_lock (&priv->lock);
if (priv->control)
- res = g_strcmp0 (priv->control, control);
+ res = (g_strcmp0 (priv->control, control) == 0);
else {
guint streamid;
- sscanf (control, "stream=%u", &streamid);
- res = (streamid == priv->idx);
+
+ if (sscanf (control, "stream=%u", &streamid) > 0)
+ res = (streamid == priv->idx);
+ else
+ res = FALSE;
}
g_mutex_unlock (&priv->lock);
}
/**
+ * gst_rtsp_stream_is_transport_supported:
+ * @stream: a #GstRTSPStream
+ * @transport: (transfer none): a #GstRTSPTransport
+ *
+ * Check if @transport can be handled by stream
+ *
+ * Returns: %TRUE if @transport can be handled by @stream.
+ */
+gboolean
+gst_rtsp_stream_is_transport_supported (GstRTSPStream * stream,
+ GstRTSPTransport * transport)
+{
+ GstRTSPStreamPrivate *priv;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ if (transport->trans != GST_RTSP_TRANS_RTP)
+ goto unsupported_transmode;
+
+ if (!(transport->profile & priv->profiles))
+ goto unsupported_profile;
+
+ if (!(transport->lower_transport & priv->protocols))
+ goto unsupported_ltrans;
+
+ g_mutex_unlock (&priv->lock);
+
+ return TRUE;
+
+ /* ERRORS */
+unsupported_transmode:
+ {
+ GST_DEBUG ("unsupported transport mode %d", transport->trans);
+ g_mutex_unlock (&priv->lock);
+ return FALSE;
+ }
+unsupported_profile:
+ {
+ GST_DEBUG ("unsupported profile %d", transport->profile);
+ g_mutex_unlock (&priv->lock);
+ return FALSE;
+ }
+unsupported_ltrans:
+ {
+ GST_DEBUG ("unsupported lower transport %d", transport->lower_transport);
+ g_mutex_unlock (&priv->lock);
+ return FALSE;
+ }
+}
+
+/**
+ * gst_rtsp_stream_set_profiles:
+ * @stream: a #GstRTSPStream
+ * @profiles: the new profiles
+ *
+ * Configure the allowed profiles for @stream.
+ */
+void
+gst_rtsp_stream_set_profiles (GstRTSPStream * stream, GstRTSPProfile profiles)
+{
+ GstRTSPStreamPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->profiles = profiles;
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_stream_get_profiles:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the allowed profiles of @stream.
+ *
+ * Returns: a #GstRTSPProfile
+ */
+GstRTSPProfile
+gst_rtsp_stream_get_profiles (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+ GstRTSPProfile res;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_RTSP_PROFILE_UNKNOWN);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ res = priv->profiles;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+/**
* gst_rtsp_stream_set_protocols:
* @stream: a #GstRTSPStream
* @protocols: the new flags
/**
* gst_rtsp_stream_set_address_pool:
* @stream: a #GstRTSPStream
- * @pool: a #GstRTSPAddressPool
+ * @pool: (transfer none): a #GstRTSPAddressPool
*
* configure @pool to be used as the address pool of @stream.
*/
*
* Get the multicast address of @stream for @family.
*
- * Returns: the #GstRTSPAddress of @stream or %NULL when no address could be
- * allocated. gst_rtsp_address_free() after usage.
+ * Returns: (transfer full) (nullable): the #GstRTSPAddress of @stream
+ * or %NULL when no address could be allocated. gst_rtsp_address_free()
+ * after usage.
*/
GstRTSPAddress *
gst_rtsp_stream_get_multicast_address (GstRTSPStream * stream,
if (family == G_SOCKET_FAMILY_IPV6) {
flags = GST_RTSP_ADDRESS_FLAG_IPV6;
- addrp = &priv->addr_v4;
+ addrp = &priv->addr_v6;
} else {
flags = GST_RTSP_ADDRESS_FLAG_IPV4;
- addrp = &priv->addr_v6;
+ addrp = &priv->addr_v4;
}
g_mutex_lock (&priv->lock);
*
* Reserve @address and @port as the address and port of @stream.
*
- * Returns: the #GstRTSPAddress of @stream or %NULL when the address could be
- * reserved. gst_rtsp_address_free() after usage.
+ * Returns: (nullable): the #GstRTSPAddress of @stream or %NULL when
+ * the address could be reserved. gst_rtsp_address_free() after usage.
*/
GstRTSPAddress *
gst_rtsp_stream_reserve_address (GstRTSPStream * stream,
}
if (family == G_SOCKET_FAMILY_IPV6)
- addrp = &priv->addr_v4;
- else
addrp = &priv->addr_v6;
+ else
+ addrp = &priv->addr_v4;
g_mutex_lock (&priv->lock);
if (*addrp == NULL) {
+ GstRTSPAddressPoolResult res;
+
if (priv->pool == NULL)
goto no_pool;
- *addrp = gst_rtsp_address_pool_reserve_address (priv->pool, address,
- port, n_ports, ttl);
- if (*addrp == NULL)
+ res = gst_rtsp_address_pool_reserve_address (priv->pool, address,
+ port, n_ports, ttl, addrp);
+ if (res != GST_RTSP_ADDRESS_POOL_OK)
goto no_address;
} else {
if (strcmp ((*addrp)->address, address) ||
g_object_set (G_OBJECT (udpsrc0), "socket", rtp_socket, NULL);
g_object_set (G_OBJECT (udpsrc1), "socket", rtcp_socket, NULL);
- ret = gst_element_set_state (udpsrc0, GST_STATE_PAUSED);
+ ret = gst_element_set_state (udpsrc0, GST_STATE_READY);
if (ret == GST_STATE_CHANGE_FAILURE)
goto element_error;
- ret = gst_element_set_state (udpsrc1, GST_STATE_PAUSED);
+ ret = gst_element_set_state (udpsrc1, GST_STATE_READY);
if (ret == GST_STATE_CHANGE_FAILURE)
goto element_error;
udpsrc_out[1] = udpsrc1;
udpsink_out[0] = udpsink0;
udpsink_out[1] = udpsink1;
+
server_port_out->min = rtpport;
server_port_out->max = rtcpport;
gst_element_set_state (udpsink0, GST_STATE_NULL);
gst_object_unref (udpsink0);
}
- if (udpsink1) {
- gst_element_set_state (udpsink1, GST_STATE_NULL);
- gst_object_unref (udpsink1);
- }
if (inetaddr)
g_object_unref (inetaddr);
g_list_free_full (rejected_addresses,
*
* Get the RTP session of this stream.
*
- * Returns: The RTP session of this stream. Unref after usage.
+ * Returns: (transfer full): The RTP session of this stream. Unref after usage.
*/
GObject *
gst_rtsp_stream_get_rtpsession (GstRTSPStream * stream)
}
}
+static void
+clear_tr_cache (GstRTSPStreamPrivate * priv)
+{
+ g_list_foreach (priv->tr_cache, (GFunc) g_object_unref, NULL);
+ g_list_free (priv->tr_cache);
+ priv->tr_cache = NULL;
+}
+
static GstFlowReturn
handle_new_sample (GstAppSink * sink, gpointer user_data)
{
GstSample *sample;
GstBuffer *buffer;
GstRTSPStream *stream;
+ gboolean is_rtp;
sample = gst_app_sink_pull_sample (sink);
if (!sample)
priv = stream->priv;
buffer = gst_sample_get_buffer (sample);
+ is_rtp = GST_ELEMENT_CAST (sink) == priv->appsink[0];
+
g_mutex_lock (&priv->lock);
- for (walk = priv->transports; walk; walk = g_list_next (walk)) {
+ if (priv->tr_cache_cookie != priv->transports_cookie) {
+ clear_tr_cache (priv);
+ for (walk = priv->transports; walk; walk = g_list_next (walk)) {
+ GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
+ priv->tr_cache = g_list_prepend (priv->tr_cache, g_object_ref (tr));
+ }
+ priv->tr_cache_cookie = priv->transports_cookie;
+ }
+ g_mutex_unlock (&priv->lock);
+
+ for (walk = priv->tr_cache; walk; walk = g_list_next (walk)) {
GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
- if (GST_ELEMENT_CAST (sink) == priv->appsink[0]) {
+ if (is_rtp) {
gst_rtsp_stream_transport_send_rtp (tr, buffer);
} else {
gst_rtsp_stream_transport_send_rtcp (tr, buffer);
}
}
- g_mutex_unlock (&priv->lock);
-
gst_sample_unref (sample);
return GST_FLOW_OK;
handle_new_sample,
};
+static GstElement *
+get_rtp_encoder (GstRTSPStream * stream, guint session)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+
+ if (priv->srtpenc == NULL) {
+ gchar *name;
+
+ name = g_strdup_printf ("srtpenc_%u", session);
+ priv->srtpenc = gst_element_factory_make ("srtpenc", name);
+ g_free (name);
+
+ g_object_set (priv->srtpenc, "random-key", TRUE, NULL);
+ }
+ return gst_object_ref (priv->srtpenc);
+}
+
+static GstElement *
+request_rtp_encoder (GstElement * rtpbin, guint session, GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+ GstElement *oldenc, *enc;
+ GstPad *pad;
+ gchar *name;
+
+ if (priv->idx != session)
+ return NULL;
+
+ GST_DEBUG_OBJECT (stream, "make RTP encoder for session %u", session);
+
+ oldenc = priv->srtpenc;
+ enc = get_rtp_encoder (stream, session);
+ name = g_strdup_printf ("rtp_sink_%d", session);
+ pad = gst_element_get_request_pad (enc, name);
+ g_free (name);
+ gst_object_unref (pad);
+
+ if (oldenc == NULL)
+ g_signal_emit (stream, gst_rtsp_stream_signals[SIGNAL_NEW_RTP_ENCODER], 0,
+ enc);
+
+ return enc;
+}
+
+static GstElement *
+request_rtcp_encoder (GstElement * rtpbin, guint session,
+ GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+ GstElement *oldenc, *enc;
+ GstPad *pad;
+ gchar *name;
+
+ if (priv->idx != session)
+ return NULL;
+
+ GST_DEBUG_OBJECT (stream, "make RTCP encoder for session %u", session);
+
+ oldenc = priv->srtpenc;
+ enc = get_rtp_encoder (stream, session);
+ name = g_strdup_printf ("rtcp_sink_%d", session);
+ pad = gst_element_get_request_pad (enc, name);
+ g_free (name);
+ gst_object_unref (pad);
+
+ if (oldenc == NULL)
+ g_signal_emit (stream, gst_rtsp_stream_signals[SIGNAL_NEW_RTCP_ENCODER], 0,
+ enc);
+
+ return enc;
+}
+
+static GstCaps *
+request_key (GstElement * srtpdec, guint ssrc, GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+ GstCaps *caps;
+
+ GST_DEBUG ("request key %08x", ssrc);
+
+ g_mutex_lock (&priv->lock);
+ if ((caps = g_hash_table_lookup (priv->keys, GINT_TO_POINTER (ssrc))))
+ gst_caps_ref (caps);
+ g_mutex_unlock (&priv->lock);
+
+ return caps;
+}
+
+static GstElement *
+request_rtcp_decoder (GstElement * rtpbin, guint session,
+ GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+
+ if (priv->idx != session)
+ return NULL;
+
+ if (priv->srtpdec == NULL) {
+ gchar *name;
+
+ name = g_strdup_printf ("srtpdec_%u", session);
+ priv->srtpdec = gst_element_factory_make ("srtpdec", name);
+ g_free (name);
+
+ g_signal_connect (priv->srtpdec, "request-key",
+ (GCallback) request_key, stream);
+ }
+ return gst_object_ref (priv->srtpdec);
+}
+
/**
* gst_rtsp_stream_join_bin:
* @stream: a #GstRTSPStream
- * @bin: a #GstBin to join
- * @rtpbin: a rtpbin element in @bin
+ * @bin: (transfer none): a #GstBin to join
+ * @rtpbin: (transfer none): a rtpbin element in @bin
* @state: the target state of the new elements
*
- * Join the #Gstbin @bin that contains the element @rtpbin.
+ * Join the #GstBin @bin that contains the element @rtpbin.
*
* @stream will link to @rtpbin, which must be inside @bin. The elements
* added to @bin will be set to the state given in @state.
/* update the dscp qos field in the sinks */
update_dscp_qos (stream);
+ if (priv->profiles & GST_RTSP_PROFILE_SAVP
+ || priv->profiles & GST_RTSP_PROFILE_SAVPF) {
+ /* For SRTP */
+ g_signal_connect (rtpbin, "request-rtp-encoder",
+ (GCallback) request_rtp_encoder, stream);
+ g_signal_connect (rtpbin, "request-rtcp-encoder",
+ (GCallback) request_rtcp_encoder, stream);
+ g_signal_connect (rtpbin, "request-rtcp-decoder",
+ (GCallback) request_rtcp_decoder, stream);
+ }
+
/* get a pad for sending RTP */
name = g_strdup_printf ("send_rtp_sink_%u", idx);
priv->send_rtp_sink = gst_element_get_request_pad (rtpbin, name);
}
/* be notified of caps changes */
- priv->caps_sig = g_signal_connect (priv->send_rtp_sink, "notify::caps",
+ priv->caps_sig = g_signal_connect (priv->send_src[0], "notify::caps",
(GCallback) caps_notify, stream);
priv->is_joined = TRUE;
/**
* gst_rtsp_stream_leave_bin:
* @stream: a #GstRTSPStream
- * @bin: a #GstBin
- * @rtpbin: a rtpbin #GstElement
+ * @bin: (transfer none): a #GstBin
+ * @rtpbin: (transfer none): a rtpbin #GstElement
*
* Remove the elements of @stream from @bin.
*
{
GstRTSPStreamPrivate *priv;
gint i;
+ GList *l;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
/* all transports must be removed by now */
g_return_val_if_fail (priv->transports == NULL, FALSE);
+ clear_tr_cache (priv);
+
GST_INFO ("stream %p leaving bin", stream);
gst_pad_unlink (priv->srcpad, priv->send_rtp_sink);
- g_signal_handler_disconnect (priv->send_rtp_sink, priv->caps_sig);
+ g_signal_handler_disconnect (priv->send_src[0], priv->caps_sig);
gst_element_release_request_pad (rtpbin, priv->send_rtp_sink);
gst_object_unref (priv->send_rtp_sink);
priv->send_rtp_sink = NULL;
gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_NULL);
gst_bin_remove (bin, priv->udpsrc_v6[i]);
}
+
+ for (l = priv->transport_sources; l; l = l->next) {
+ GstRTSPMulticastTransportSource *s = l->data;
+
+ if (!s->udpsrc[i])
+ continue;
+
+ gst_element_set_locked_state (s->udpsrc[i], FALSE);
+ gst_element_set_state (s->udpsrc[i], GST_STATE_NULL);
+ gst_bin_remove (bin, s->udpsrc[i]);
+ }
+
if (priv->udpsink[i])
gst_bin_remove (bin, priv->udpsink[i]);
if (priv->appsrc[i])
priv->tee[i] = NULL;
priv->funnel[i] = NULL;
}
+
+ for (l = priv->transport_sources; l; l = l->next) {
+ GstRTSPMulticastTransportSource *s = l->data;
+ g_slice_free (GstRTSPMulticastTransportSource, s);
+ }
+ g_list_free (priv->transport_sources);
+ priv->transport_sources = NULL;
+
gst_object_unref (priv->send_src[0]);
priv->send_src[0] = NULL;
gst_caps_unref (priv->caps);
priv->caps = NULL;
+ if (priv->srtpenc)
+ gst_object_unref (priv->srtpenc);
+
priv->is_joined = FALSE;
g_mutex_unlock (&priv->lock);
was_not_joined:
{
+ g_mutex_unlock (&priv->lock);
return TRUE;
}
}
/**
* gst_rtsp_stream_get_rtpinfo:
* @stream: a #GstRTSPStream
- * @rtptime: result RTP timestamp
- * @seq: result RTP seqnum
+ * @rtptime: (allow-none): result RTP timestamp
+ * @seq: (allow-none): result RTP seqnum
+ * @clock_rate: (allow-none): the clock rate
+ * @running_time: (allow-none): result running-time
*
- * Retrieve the current rtptime and seq. This is used to
+ * Retrieve the current rtptime, seq and running-time. This is used to
* construct a RTPInfo reply header.
*
- * Returns: %TRUE when rtptime and seq could be determined.
+ * Returns: %TRUE when rtptime, seq and running-time could be determined.
*/
gboolean
gst_rtsp_stream_get_rtpinfo (GstRTSPStream * stream,
- guint * rtptime, guint * seq)
+ guint * rtptime, guint * seq, guint * clock_rate,
+ GstClockTime * running_time)
{
GstRTSPStreamPrivate *priv;
+ GstStructure *stats;
GObjectClass *payobjclass;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
- g_return_val_if_fail (rtptime != NULL, FALSE);
- g_return_val_if_fail (seq != NULL, FALSE);
priv = stream->priv;
payobjclass = G_OBJECT_GET_CLASS (priv->payloader);
- if (!g_object_class_find_property (payobjclass, "seqnum") ||
- !g_object_class_find_property (payobjclass, "timestamp"))
- return FALSE;
+ g_mutex_lock (&priv->lock);
+
+ if (g_object_class_find_property (payobjclass, "stats")) {
+ g_object_get (priv->payloader, "stats", &stats, NULL);
+ if (stats == NULL)
+ goto no_stats;
+
+ if (seq)
+ gst_structure_get_uint (stats, "seqnum", seq);
+
+ if (rtptime)
+ gst_structure_get_uint (stats, "timestamp", rtptime);
+
+ if (running_time)
+ gst_structure_get_clock_time (stats, "running-time", running_time);
+
+ if (clock_rate) {
+ gst_structure_get_uint (stats, "clock-rate", clock_rate);
+ if (*clock_rate == 0 && running_time)
+ *running_time = GST_CLOCK_TIME_NONE;
+ }
+ gst_structure_free (stats);
+ } else {
+ if (!g_object_class_find_property (payobjclass, "seqnum") ||
+ !g_object_class_find_property (payobjclass, "timestamp"))
+ goto no_stats;
- g_object_get (priv->payloader, "seqnum", seq, "timestamp", rtptime, NULL);
+ if (seq)
+ g_object_get (priv->payloader, "seqnum", seq, NULL);
+
+ if (rtptime)
+ g_object_get (priv->payloader, "timestamp", rtptime, NULL);
+
+ if (running_time)
+ *running_time = GST_CLOCK_TIME_NONE;
+ }
+ g_mutex_unlock (&priv->lock);
return TRUE;
+
+ /* ERRORS */
+no_stats:
+ {
+ GST_WARNING ("Could not get payloader stats");
+ g_mutex_unlock (&priv->lock);
+ return FALSE;
+ }
}
/**
* Retrieve the current caps of @stream.
*
* Returns: (transfer full): the #GstCaps of @stream. use gst_caps_unref()
- * after usage.
+ * after usage.
*/
GstCaps *
gst_rtsp_stream_get_caps (GstRTSPStream * stream)
gst_object_unref (element);
} else {
ret = GST_FLOW_OK;
+ gst_buffer_unref (buffer);
}
return ret;
}
tr = gst_rtsp_stream_transport_get_transport (trans);
switch (tr->lower_transport) {
- case GST_RTSP_LOWER_TRANS_UDP:
case GST_RTSP_LOWER_TRANS_UDP_MCAST:
{
+ GstRTSPMulticastTransportSource *source;
+ GstBin *bin;
+
+ bin = GST_BIN (gst_object_get_parent (GST_OBJECT (priv->funnel[0])));
+
+ if (add) {
+ gchar *host;
+ gint i;
+ GstPad *selpad, *pad;
+
+ source = g_slice_new0 (GstRTSPMulticastTransportSource);
+ source->transport = trans;
+
+ for (i = 0; i < 2; i++) {
+ host =
+ g_strdup_printf ("udp://%s:%d", tr->destination,
+ (i == 0) ? tr->port.min : tr->port.max);
+ source->udpsrc[i] =
+ gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
+ g_free (host);
+
+ /* we set and keep these to playing so that they don't cause NO_PREROLL return
+ * values */
+ gst_element_set_state (source->udpsrc[i], GST_STATE_PLAYING);
+ gst_element_set_locked_state (source->udpsrc[i], TRUE);
+ /* add udpsrc */
+ gst_bin_add (bin, source->udpsrc[i]);
+
+ /* and link to the funnel v4 */
+ source->selpad[i] = selpad =
+ gst_element_get_request_pad (priv->funnel[i], "sink_%u");
+ pad = gst_element_get_static_pad (source->udpsrc[i], "src");
+ gst_pad_link (pad, selpad);
+ gst_object_unref (pad);
+ gst_object_unref (selpad);
+ }
+ gst_object_unref (bin);
+
+ priv->transport_sources =
+ g_list_prepend (priv->transport_sources, source);
+ } else {
+ GList *l;
+
+ for (l = priv->transport_sources; l; l = l->next) {
+ source = l->data;
+
+ if (source->transport == trans) {
+ priv->transport_sources =
+ g_list_delete_link (priv->transport_sources, l);
+ break;
+ }
+ }
+
+ if (l != NULL) {
+ gint i;
+
+ for (i = 0; i < 2; i++) {
+ /* Will automatically unlink everything */
+ gst_bin_remove (bin,
+ GST_ELEMENT (gst_object_ref (source->udpsrc[i])));
+
+ gst_element_set_state (source->udpsrc[i], GST_STATE_NULL);
+ gst_object_unref (source->udpsrc[i]);
+
+ gst_element_release_request_pad (priv->funnel[i],
+ source->selpad[i]);
+ }
+
+ g_slice_free (GstRTSPMulticastTransportSource, source);
+ }
+ }
+
+ /* fall through for the generic case */
+ }
+ case GST_RTSP_LOWER_TRANS_UDP:
+ {
gchar *dest;
gint min, max;
guint ttl = 0;
}
if (add) {
- GST_INFO ("adding %s:%d-%d", dest, min, max);
- g_signal_emit_by_name (priv->udpsink[0], "add", dest, min, NULL);
- g_signal_emit_by_name (priv->udpsink[1], "add", dest, max, NULL);
if (ttl > 0) {
GST_INFO ("setting ttl-mc %d", ttl);
g_object_set (G_OBJECT (priv->udpsink[0]), "ttl-mc", ttl, NULL);
g_object_set (G_OBJECT (priv->udpsink[1]), "ttl-mc", ttl, NULL);
}
+ GST_INFO ("adding %s:%d-%d", dest, min, max);
+ g_signal_emit_by_name (priv->udpsink[0], "add", dest, min, NULL);
+ g_signal_emit_by_name (priv->udpsink[1], "add", dest, max, NULL);
priv->transports = g_list_prepend (priv->transports, trans);
} else {
GST_INFO ("removing %s:%d-%d", dest, min, max);
g_signal_emit_by_name (priv->udpsink[1], "remove", dest, max, NULL);
priv->transports = g_list_remove (priv->transports, trans);
}
+ priv->transports_cookie++;
break;
}
case GST_RTSP_LOWER_TRANS_TCP:
GST_INFO ("removing TCP %s", tr->destination);
priv->transports = g_list_remove (priv->transports, trans);
}
+ priv->transports_cookie++;
break;
default:
goto unknown_transport;
/**
* gst_rtsp_stream_add_transport:
* @stream: a #GstRTSPStream
- * @trans: a #GstRTSPStreamTransport
+ * @trans: (transfer none): a #GstRTSPStreamTransport
*
* Add the transport in @trans to @stream. The media of @stream will
* then also be send to the values configured in @trans.
/**
* gst_rtsp_stream_remove_transport:
* @stream: a #GstRTSPStream
- * @trans: a #GstRTSPStreamTransport
+ * @trans: (transfer none): a #GstRTSPStreamTransport
*
* Remove the transport in @trans from @stream. The media of @stream will
* not be sent to the values configured in @trans.
return res;
}
+
+/**
+ * gst_rtsp_stream_update_crypto:
+ * @stream: a #GstRTSPStream
+ * @ssrc: the SSRC
+ * @crypto: (transfer none) (allow-none): a #GstCaps with crypto info
+ *
+ * Update the new crypto information for @ssrc in @stream. If information
+ * for @ssrc did not exist, it will be added. If information
+ * for @ssrc existed, it will be replaced. If @crypto is %NULL, it will
+ * be removed from @stream.
+ *
+ * Returns: %TRUE if @crypto could be updated
+ */
+gboolean
+gst_rtsp_stream_update_crypto (GstRTSPStream * stream,
+ guint ssrc, GstCaps * crypto)
+{
+ GstRTSPStreamPrivate *priv;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+ g_return_val_if_fail (crypto == NULL || GST_IS_CAPS (crypto), FALSE);
+
+ priv = stream->priv;
+
+ GST_DEBUG_OBJECT (stream, "update key for %08x", ssrc);
+
+ g_mutex_lock (&priv->lock);
+ if (crypto)
+ g_hash_table_insert (priv->keys, GINT_TO_POINTER (ssrc),
+ gst_caps_ref (crypto));
+ else
+ g_hash_table_remove (priv->keys, GINT_TO_POINTER (ssrc));
+ g_mutex_unlock (&priv->lock);
+
+ return TRUE;
+}
+
+/**
+ * gst_rtsp_stream_get_rtp_socket:
+ * @stream: a #GstRTSPStream
+ * @family: the socket family
+ *
+ * Get the RTP socket from @stream for a @family.
+ *
+ * @stream must be joined to a bin.
+ *
+ * Returns: (transfer full) (nullable): the RTP socket or %NULL if no
+ * socket could be allocated for @family. Unref after usage
+ */
+GSocket *
+gst_rtsp_stream_get_rtp_socket (GstRTSPStream * stream, GSocketFamily family)
+{
+ GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
+ GSocket *socket;
+ const gchar *name;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+ g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
+ family == G_SOCKET_FAMILY_IPV6, NULL);
+ g_return_val_if_fail (priv->udpsink[0], NULL);
+
+ if (family == G_SOCKET_FAMILY_IPV6)
+ name = "socket-v6";
+ else
+ name = "socket";
+
+ g_object_get (priv->udpsink[0], name, &socket, NULL);
+
+ return socket;
+}
+
+/**
+ * gst_rtsp_stream_get_rtcp_socket:
+ * @stream: a #GstRTSPStream
+ * @family: the socket family
+ *
+ * Get the RTCP socket from @stream for a @family.
+ *
+ * @stream must be joined to a bin.
+ *
+ * Returns: (transfer full) (nullable): the RTCP socket or %NULL if no
+ * socket could be allocated for @family. Unref after usage
+ */
+GSocket *
+gst_rtsp_stream_get_rtcp_socket (GstRTSPStream * stream, GSocketFamily family)
+{
+ GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
+ GSocket *socket;
+ const gchar *name;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+ g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
+ family == G_SOCKET_FAMILY_IPV6, NULL);
+ g_return_val_if_fail (priv->udpsink[1], NULL);
+
+ if (family == G_SOCKET_FAMILY_IPV6)
+ name = "socket-v6";
+ else
+ name = "socket";
+
+ g_object_get (priv->udpsink[1], name, &socket, NULL);
+
+ return socket;
+}
+
+/**
+ * gst_rtsp_stream_transport_filter:
+ * @stream: a #GstRTSPStream
+ * @func: (scope call) (allow-none): a callback
+ * @user_data: (closure): user data passed to @func
+ *
+ * Call @func for each transport managed by @stream. The result value of @func
+ * determines what happens to the transport. @func will be called with @stream
+ * locked so no further actions on @stream can be performed from @func.
+ *
+ * If @func returns #GST_RTSP_FILTER_REMOVE, the transport will be removed from
+ * @stream.
+ *
+ * If @func returns #GST_RTSP_FILTER_KEEP, the transport will remain in @stream.
+ *
+ * If @func returns #GST_RTSP_FILTER_REF, the transport will remain in @stream but
+ * will also be added with an additional ref to the result #GList of this
+ * function..
+ *
+ * When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for each transport.
+ *
+ * Returns: (element-type GstRTSPStreamTransport) (transfer full): a #GList with all
+ * transports for which @func returned #GST_RTSP_FILTER_REF. After usage, each
+ * element in the #GList should be unreffed before the list is freed.
+ */
+GList *
+gst_rtsp_stream_transport_filter (GstRTSPStream * stream,
+ GstRTSPStreamTransportFilterFunc func, gpointer user_data)
+{
+ GstRTSPStreamPrivate *priv;
+ GList *result, *walk, *next;
+ GHashTable *visited;
+ guint cookie;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+
+ priv = stream->priv;
+
+ result = NULL;
+ if (func)
+ visited = g_hash_table_new_full (NULL, NULL, g_object_unref, NULL);
+
+ g_mutex_lock (&priv->lock);
+restart:
+ cookie = priv->transports_cookie;
+ for (walk = priv->transports; walk; walk = next) {
+ GstRTSPStreamTransport *trans = walk->data;
+ GstRTSPFilterResult res;
+ gboolean changed;
+
+ next = g_list_next (walk);
+
+ if (func) {
+ /* only visit each transport once */
+ if (g_hash_table_contains (visited, trans))
+ continue;
+
+ g_hash_table_add (visited, g_object_ref (trans));
+ g_mutex_unlock (&priv->lock);
+
+ res = func (stream, trans, user_data);
+
+ g_mutex_lock (&priv->lock);
+ } else
+ res = GST_RTSP_FILTER_REF;
+
+ changed = (cookie != priv->transports_cookie);
+
+ switch (res) {
+ case GST_RTSP_FILTER_REMOVE:
+ update_transport (stream, trans, FALSE);
+ break;
+ case GST_RTSP_FILTER_REF:
+ result = g_list_prepend (result, g_object_ref (trans));
+ break;
+ case GST_RTSP_FILTER_KEEP:
+ default:
+ break;
+ }
+ if (changed)
+ goto restart;
+ }
+ g_mutex_unlock (&priv->lock);
+
+ if (func)
+ g_hash_table_unref (visited);
+
+ return result;
+}
+
+static GstPadProbeReturn
+pad_blocking (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
+{
+ GstRTSPStreamPrivate *priv;
+ GstRTSPStream *stream;
+
+ stream = user_data;
+ priv = stream->priv;
+
+ GST_DEBUG_OBJECT (pad, "now blocking");
+
+ g_mutex_lock (&priv->lock);
+ priv->blocking = TRUE;
+ g_mutex_unlock (&priv->lock);
+
+ gst_element_post_message (priv->payloader,
+ gst_message_new_element (GST_OBJECT_CAST (priv->payloader),
+ gst_structure_new_empty ("GstRTSPStreamBlocking")));
+
+ return GST_PAD_PROBE_OK;
+}
+
+/**
+ * gst_rtsp_stream_set_blocked:
+ * @stream: a #GstRTSPStream
+ * @blocked: boolean indicating we should block or unblock
+ *
+ * Blocks or unblocks the dataflow on @stream.
+ *
+ * Returns: %TRUE on success
+ */
+gboolean
+gst_rtsp_stream_set_blocked (GstRTSPStream * stream, gboolean blocked)
+{
+ GstRTSPStreamPrivate *priv;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ if (blocked) {
+ priv->blocking = FALSE;
+ if (priv->blocked_id == 0) {
+ priv->blocked_id = gst_pad_add_probe (priv->srcpad,
+ GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER |
+ GST_PAD_PROBE_TYPE_BUFFER_LIST, pad_blocking,
+ g_object_ref (stream), g_object_unref);
+ }
+ } else {
+ if (priv->blocked_id != 0) {
+ gst_pad_remove_probe (priv->srcpad, priv->blocked_id);
+ priv->blocked_id = 0;
+ priv->blocking = FALSE;
+ }
+ }
+ g_mutex_unlock (&priv->lock);
+
+ return TRUE;
+}
+
+/**
+ * gst_rtsp_stream_is_blocking:
+ * @stream: a #GstRTSPStream
+ *
+ * Check if @stream is blocking on a #GstBuffer.
+ *
+ * Returns: %TRUE if @stream is blocking
+ */
+gboolean
+gst_rtsp_stream_is_blocking (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+ gboolean result;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ result = priv->blocking;
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_stream_query_position:
+ * @stream: a #GstRTSPStream
+ *
+ * Query the position of the stream in %GST_FORMAT_TIME. This only considers
+ * the RTP parts of the pipeline and not the RTCP parts.
+ *
+ * Returns: %TRUE if the position could be queried
+ */
+gboolean
+gst_rtsp_stream_query_position (GstRTSPStream * stream, gint64 * position)
+{
+ GstRTSPStreamPrivate *priv;
+ GstElement *sink;
+ gboolean ret;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ if ((sink = priv->udpsink[0]))
+ gst_object_ref (sink);
+ g_mutex_unlock (&priv->lock);
+
+ if (!sink)
+ return FALSE;
+
+ ret = gst_element_query_position (sink, GST_FORMAT_TIME, position);
+ gst_object_unref (sink);
+
+ return ret;
+}
+
+/**
+ * gst_rtsp_stream_query_stop:
+ * @stream: a #GstRTSPStream
+ *
+ * Query the stop of the stream in %GST_FORMAT_TIME. This only considers
+ * the RTP parts of the pipeline and not the RTCP parts.
+ *
+ * Returns: %TRUE if the stop could be queried
+ */
+gboolean
+gst_rtsp_stream_query_stop (GstRTSPStream * stream, gint64 * stop)
+{
+ GstRTSPStreamPrivate *priv;
+ GstElement *sink;
+ GstQuery *query;
+ gboolean ret;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ if ((sink = priv->udpsink[0]))
+ gst_object_ref (sink);
+ g_mutex_unlock (&priv->lock);
+
+ if (!sink)
+ return FALSE;
+
+ query = gst_query_new_segment (GST_FORMAT_TIME);
+ if ((ret = gst_element_query (sink, query))) {
+ GstFormat format;
+
+ gst_query_parse_segment (query, NULL, &format, NULL, stop);
+ if (format != GST_FORMAT_TIME)
+ *stop = -1;
+ }
+ gst_query_unref (query);
+ gst_object_unref (sink);
+
+ return ret;
+
+}
projects / platform / upstream / gstreamer.git / blobdiff