*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
*/
/*
* Unless otherwise indicated, Source Code is licensed under MIT license.
* <refsect2>
* <title>Example launch line</title>
* |[
- * gst-launch rtspsrc location=rtsp://some.server/url ! fakesink
+ * gst-launch-1.0 rtspsrc location=rtsp://some.server/url ! fakesink
* ]| Establish a connection to an RTSP server and send the raw RTP packets to a
* fakesink.
* </refsect2>
#endif /* HAVE_UNISTD_H */
#include <stdlib.h>
#include <string.h>
-#include <locale.h>
#include <stdio.h>
#include <stdarg.h>
+#include <gst/net/gstnet.h>
#include <gst/sdp/gstsdpmessage.h>
#include <gst/rtp/gstrtppayloads.h>
enum
{
- /* FILL ME */
+ SIGNAL_HANDLE_REQUEST,
LAST_SIGNAL
};
#define DEFAULT_UDP_BUFFER_SIZE 0x80000
#define DEFAULT_TCP_TIMEOUT 20000000
#define DEFAULT_LATENCY_MS 2000
+#define DEFAULT_DROP_ON_LATENCY FALSE
#define DEFAULT_CONNECTION_SPEED 0
#define DEFAULT_NAT_METHOD GST_RTSP_NAT_DUMMY
#define DEFAULT_DO_RTCP TRUE
#define DEFAULT_BUFFER_MODE BUFFER_MODE_AUTO
#define DEFAULT_PORT_RANGE NULL
#define DEFAULT_SHORT_HEADER FALSE
+#define DEFAULT_PROBATION 2
+#define DEFAULT_UDP_RECONNECT TRUE
+#define DEFAULT_MULTICAST_IFACE NULL
+#define DEFAULT_NTP_SYNC FALSE
+#define DEFAULT_USE_PIPELINE_CLOCK FALSE
enum
{
PROP_TIMEOUT,
PROP_TCP_TIMEOUT,
PROP_LATENCY,
+ PROP_DROP_ON_LATENCY,
PROP_CONNECTION_SPEED,
PROP_NAT_METHOD,
PROP_DO_RTCP,
PROP_DO_RTSP_KEEP_ALIVE,
PROP_PROXY,
+ PROP_PROXY_ID,
+ PROP_PROXY_PW,
PROP_RTP_BLOCKSIZE,
PROP_USER_ID,
PROP_USER_PW,
PROP_PORT_RANGE,
PROP_UDP_BUFFER_SIZE,
PROP_SHORT_HEADER,
+ PROP_PROBATION,
+ PROP_UDP_RECONNECT,
+ PROP_MULTICAST_IFACE,
+ PROP_NTP_SYNC,
+ PROP_USE_PIPELINE_CLOCK,
PROP_LAST
};
static void gst_rtspsrc_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
+static GstClock *gst_rtspsrc_provide_clock (GstElement * element);
+
static void gst_rtspsrc_uri_handler_init (gpointer g_iface,
gpointer iface_data);
static gboolean gst_rtspsrc_uri_set_uri (GstURIHandler * handler,
const gchar * uri, GError ** error);
+static gchar *gst_rtspsrc_uri_get_uri (GstURIHandler * handler);
static gboolean gst_rtspsrc_activate_streams (GstRTSPSrc * src);
static gboolean gst_rtspsrc_loop (GstRTSPSrc * src);
#define CMD_RECONNECT (1 << 5)
#define CMD_LOOP (1 << 6)
+/* mask for all commands */
+#define CMD_ALL ((CMD_LOOP << 1) - 1)
+
#define GST_ELEMENT_PROGRESS(el, type, code, text) \
G_STMT_START { \
gchar *__txt = _gst_element_error_printf text; \
g_free (__txt); \
} G_STMT_END
-/*static guint gst_rtspsrc_signals[LAST_SIGNAL] = { 0 }; */
+static guint gst_rtspsrc_signals[LAST_SIGNAL] = { 0 };
+
#define gst_rtspsrc_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (GstRTSPSrc, gst_rtspsrc, GST_TYPE_BIN,
G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER, gst_rtspsrc_uri_handler_init));
"Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
+ g_param_spec_boolean ("drop-on-latency",
+ "Drop buffers when maximum latency is reached",
+ "Tells the jitterbuffer to never exceed the given latency in size",
+ DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
g_object_class_install_property (gobject_class, PROP_CONNECTION_SPEED,
g_param_spec_uint64 ("connection-speed", "Connection Speed",
"Network connection speed in kbps (0 = unknown)",
g_param_spec_string ("proxy", "Proxy",
"Proxy settings for HTTP tunneling. Format: [http://][user:passwd@]host[:port]",
DEFAULT_PROXY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ /**
+ * GstRTSPSrc::proxy-id
+ *
+ * Sets the proxy URI user id for authentication. If the URI set via the
+ * "proxy" property contains a user-id already, that will take precedence.
+ *
+ * Since: 1.2
+ */
+ g_object_class_install_property (gobject_class, PROP_PROXY_ID,
+ g_param_spec_string ("proxy-id", "proxy-id",
+ "HTTP proxy URI user id for authentication", "",
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ /**
+ * GstRTSPSrc::proxy-pw
+ *
+ * Sets the proxy URI password for authentication. If the URI set via the
+ * "proxy" property contains a password already, that will take precedence.
+ *
+ * Since: 1.2
+ */
+ g_object_class_install_property (gobject_class, PROP_PROXY_PW,
+ g_param_spec_string ("proxy-pw", "proxy-pw",
+ "HTTP proxy URI user password for authentication", "",
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRTSPSrc::rtp_blocksize
"Only send the basic RTSP headers for broken encoders",
DEFAULT_SHORT_HEADER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ g_object_class_install_property (gobject_class, PROP_PROBATION,
+ g_param_spec_uint ("probation", "Number of probations",
+ "Consecutive packet sequence numbers to accept the source",
+ 0, G_MAXUINT, DEFAULT_PROBATION,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_UDP_RECONNECT,
+ g_param_spec_boolean ("udp-reconnect", "Reconnect to the server",
+ "Reconnect to the server if RTSP connection is closed when doing UDP",
+ DEFAULT_UDP_RECONNECT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_MULTICAST_IFACE,
+ g_param_spec_string ("multicast-iface", "Multicast Interface",
+ "The network interface on which to join the multicast group",
+ DEFAULT_MULTICAST_IFACE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_NTP_SYNC,
+ g_param_spec_boolean ("ntp-sync", "Sync on NTP clock",
+ "Synchronize received streams to the NTP clock", DEFAULT_NTP_SYNC,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_USE_PIPELINE_CLOCK,
+ g_param_spec_boolean ("use-pipeline-clock", "Use pipeline clock",
+ "Use the pipeline running-time to set the NTP time in the RTCP SR messages",
+ DEFAULT_USE_PIPELINE_CLOCK,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ /**
+ * GstRTSPSrc::handle-request:
+ * @rtspsrc: a #GstRTSPSrc
+ * @request: a #GstRTSPMessage
+ * @response: a #GstRTSPMessage
+ *
+ * Handle a server request in @request and prepare @response.
+ *
+ * This signal is called from the streaming thread, you should therefore not
+ * do any state changes on @rtspsrc because this might deadlock. If you want
+ * to modify the state as a result of this signal, post a
+ * #GST_MESSAGE_REQUEST_STATE message on the bus or signal the main thread
+ * in some other way.
+ *
+ * Since: 1.2
+ */
+ gst_rtspsrc_signals[SIGNAL_HANDLE_REQUEST] =
+ g_signal_new ("handle-request", G_TYPE_FROM_CLASS (klass), 0,
+ 0, NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2,
+ G_TYPE_POINTER, G_TYPE_POINTER);
+
gstelement_class->send_event = gst_rtspsrc_send_event;
+ gstelement_class->provide_clock = gst_rtspsrc_provide_clock;
gstelement_class->change_state = gst_rtspsrc_change_state;
gst_element_class_add_pad_template (gstelement_class,
src->udp_timeout = DEFAULT_TIMEOUT;
gst_rtspsrc_set_tcp_timeout (src, DEFAULT_TCP_TIMEOUT);
src->latency = DEFAULT_LATENCY_MS;
+ src->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
src->connection_speed = DEFAULT_CONNECTION_SPEED;
src->nat_method = DEFAULT_NAT_METHOD;
src->do_rtcp = DEFAULT_DO_RTCP;
src->client_port_range.max = 0;
src->udp_buffer_size = DEFAULT_UDP_BUFFER_SIZE;
src->short_header = DEFAULT_SHORT_HEADER;
+ src->probation = DEFAULT_PROBATION;
+ src->udp_reconnect = DEFAULT_UDP_RECONNECT;
+ src->multi_iface = g_strdup (DEFAULT_MULTICAST_IFACE);
+ src->ntp_sync = DEFAULT_NTP_SYNC;
+ src->use_pipeline_clock = DEFAULT_USE_PIPELINE_CLOCK;
/* get a list of all extensions */
src->extensions = gst_rtsp_ext_list_get ();
g_free (rtspsrc->conninfo.url_str);
g_free (rtspsrc->user_id);
g_free (rtspsrc->user_pw);
+ g_free (rtspsrc->multi_iface);
if (rtspsrc->sdp) {
gst_sdp_message_free (rtspsrc->sdp);
rtspsrc->sdp = NULL;
}
+ if (rtspsrc->provided_clock)
+ gst_object_unref (rtspsrc->provided_clock);
/* free locks */
g_rec_mutex_clear (&rtspsrc->stream_rec_lock);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
+static GstClock *
+gst_rtspsrc_provide_clock (GstElement * element)
+{
+ GstRTSPSrc *src = GST_RTSPSRC (element);
+ GstClock *clock;
+
+ if ((clock = src->provided_clock) != NULL)
+ gst_object_ref (clock);
+
+ return clock;
+}
+
/* a proxy string of the format [user:passwd@]host[:port] */
static gboolean
gst_rtspsrc_set_proxy (GstRTSPSrc * rtsp, const gchar * proxy)
/* move to host */
p = at + 1;
+ } else {
+ if (rtsp->prop_proxy_id != NULL && *rtsp->prop_proxy_id != '\0')
+ rtsp->proxy_user = g_strdup (rtsp->prop_proxy_id);
+ if (rtsp->prop_proxy_pw != NULL && *rtsp->prop_proxy_pw != '\0')
+ rtsp->proxy_passwd = g_strdup (rtsp->prop_proxy_pw);
+ if (rtsp->proxy_user != NULL || rtsp->proxy_passwd != NULL) {
+ GST_LOG_OBJECT (rtsp, "set proxy user/pw from properties: %s:%s",
+ GST_STR_NULL (rtsp->proxy_user), GST_STR_NULL (rtsp->proxy_passwd));
+ }
}
col = strchr (p, ':');
case PROP_LATENCY:
rtspsrc->latency = g_value_get_uint (value);
break;
+ case PROP_DROP_ON_LATENCY:
+ rtspsrc->drop_on_latency = g_value_get_boolean (value);
+ break;
case PROP_CONNECTION_SPEED:
rtspsrc->connection_speed = g_value_get_uint64 (value);
break;
case PROP_PROXY:
gst_rtspsrc_set_proxy (rtspsrc, g_value_get_string (value));
break;
+ case PROP_PROXY_ID:
+ if (rtspsrc->prop_proxy_id)
+ g_free (rtspsrc->prop_proxy_id);
+ rtspsrc->prop_proxy_id = g_value_dup_string (value);
+ break;
+ case PROP_PROXY_PW:
+ if (rtspsrc->prop_proxy_pw)
+ g_free (rtspsrc->prop_proxy_pw);
+ rtspsrc->prop_proxy_pw = g_value_dup_string (value);
+ break;
case PROP_RTP_BLOCKSIZE:
rtspsrc->rtp_blocksize = g_value_get_uint (value);
break;
case PROP_SHORT_HEADER:
rtspsrc->short_header = g_value_get_boolean (value);
break;
+ case PROP_PROBATION:
+ rtspsrc->probation = g_value_get_uint (value);
+ break;
+ case PROP_UDP_RECONNECT:
+ rtspsrc->udp_reconnect = g_value_get_boolean (value);
+ break;
+ case PROP_MULTICAST_IFACE:
+ g_free (rtspsrc->multi_iface);
+
+ if (g_value_get_string (value) == NULL)
+ rtspsrc->multi_iface = g_strdup (DEFAULT_MULTICAST_IFACE);
+ else
+ rtspsrc->multi_iface = g_value_dup_string (value);
+ break;
+ case PROP_NTP_SYNC:
+ rtspsrc->ntp_sync = g_value_get_boolean (value);
+ break;
+ case PROP_USE_PIPELINE_CLOCK:
+ rtspsrc->use_pipeline_clock = g_value_get_boolean (value);
+ break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
case PROP_LATENCY:
g_value_set_uint (value, rtspsrc->latency);
break;
+ case PROP_DROP_ON_LATENCY:
+ g_value_set_boolean (value, rtspsrc->drop_on_latency);
+ break;
case PROP_CONNECTION_SPEED:
g_value_set_uint64 (value, rtspsrc->connection_speed);
break;
g_value_take_string (value, str);
break;
}
+ case PROP_PROXY_ID:
+ g_value_set_string (value, rtspsrc->prop_proxy_id);
+ break;
+ case PROP_PROXY_PW:
+ g_value_set_string (value, rtspsrc->prop_proxy_pw);
+ break;
case PROP_RTP_BLOCKSIZE:
g_value_set_uint (value, rtspsrc->rtp_blocksize);
break;
case PROP_SHORT_HEADER:
g_value_set_boolean (value, rtspsrc->short_header);
break;
+ case PROP_PROBATION:
+ g_value_set_uint (value, rtspsrc->probation);
+ break;
+ case PROP_UDP_RECONNECT:
+ g_value_set_boolean (value, rtspsrc->udp_reconnect);
+ break;
+ case PROP_MULTICAST_IFACE:
+ g_value_set_string (value, rtspsrc->multi_iface);
+ break;
+ case PROP_NTP_SYNC:
+ g_value_set_boolean (value, rtspsrc->ntp_sync);
+ break;
+ case PROP_USE_PIPELINE_CLOCK:
+ g_value_set_boolean (value, rtspsrc->use_pipeline_clock);
+ break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
gst_sdp_message_free (src->sdp);
src->sdp = NULL;
}
+ if (src->start_segment) {
+ gst_event_unref (src->start_segment);
+ src->start_segment = NULL;
+ }
+ if (src->provided_clock) {
+ gst_object_unref (src->provided_clock);
+ src->provided_clock = NULL;
+ }
}
#define PARSE_INT(p, del, res) \
GST_DEBUG_OBJECT (src, "free RTP udpsrc");
gst_element_set_state (udpsrc0, GST_STATE_NULL);
gst_object_unref (udpsrc0);
+ udpsrc0 = NULL;
GST_DEBUG_OBJECT (src, "retry %d", count);
goto again;
GST_DEBUG_OBJECT (src, "free RTP udpsrc");
gst_element_set_state (udpsrc0, GST_STATE_NULL);
gst_object_unref (udpsrc0);
+ udpsrc0 = NULL;
GST_DEBUG_OBJECT (src, "retry %d", count);
tmp_rtp++;
/* set port */
tmp_rtcp = tmp_rtp + 1;
- if (src->client_port_range.max > 0 && tmp_rtcp >= src->client_port_range.max)
+ if (src->client_port_range.max > 0 && tmp_rtcp > src->client_port_range.max)
goto no_ports;
g_object_set (G_OBJECT (udpsrc1), "port", tmp_rtcp, "reuse", FALSE, NULL);
GST_DEBUG_OBJECT (src, "free RTP udpsrc");
gst_element_set_state (udpsrc0, GST_STATE_NULL);
gst_object_unref (udpsrc0);
+ udpsrc0 = NULL;
GST_DEBUG_OBJECT (src, "free RTCP udpsrc");
gst_element_set_state (udpsrc1, GST_STATE_NULL);
* server told us to really use the UDP ports. */
stream->udpsrc[0] = gst_object_ref_sink (udpsrc0);
stream->udpsrc[1] = gst_object_ref_sink (udpsrc1);
+ gst_element_set_locked_state (stream->udpsrc[0], TRUE);
+ gst_element_set_locked_state (stream->udpsrc[1], TRUE);
/* keep track of next available port number when we have a range
* configured */
}
static void
+gst_rtspsrc_set_state (GstRTSPSrc * src, GstState state)
+{
+ GList *walk;
+
+ if (src->manager)
+ gst_element_set_state (GST_ELEMENT_CAST (src->manager), state);
+
+ for (walk = src->streams; walk; walk = g_list_next (walk)) {
+ GstRTSPStream *stream = (GstRTSPStream *) walk->data;
+ gint i;
+
+ for (i = 0; i < 2; i++) {
+ if (stream->udpsrc[i])
+ gst_element_set_state (stream->udpsrc[i], state);
+ }
+ }
+}
+
+static void
gst_rtspsrc_flush (GstRTSPSrc * src, gboolean flush, gboolean playing)
{
GstEvent *event;
- gint cmd, i;
+ gint cmd;
GstState state;
- GList *walk;
if (flush) {
event = gst_event_new_flush_start ();
}
gst_rtspsrc_push_event (src, event);
gst_rtspsrc_loop_send_cmd (src, cmd, CMD_LOOP);
-
- /* to manage jitterbuffer buffer mode */
- if (src->manager)
- gst_element_set_state (GST_ELEMENT_CAST (src->manager), state);
-
- /* make running time start start at 0 again */
- for (walk = src->streams; walk; walk = g_list_next (walk)) {
- GstRTSPStream *stream = (GstRTSPStream *) walk->data;
-
- for (i = 0; i < 2; i++) {
- /* for udp case */
- if (stream->udpsrc[i]) {
- gst_element_set_state (stream->udpsrc[i], state);
- }
- }
- }
+ gst_rtspsrc_set_state (src, state);
}
static GstRTSPResult
GST_DEBUG_OBJECT (src, "Creating newsegment from %" G_GINT64_FORMAT
" to %" G_GINT64_FORMAT, src->segment.position, stop);
- /* store the newsegment event so it can be sent from the streaming thread. */
- if (src->start_segment)
- gst_event_unref (src->start_segment);
- src->start_segment = gst_event_new_segment (&src->segment);
-
/* mark discont */
GST_DEBUG_OBJECT (src, "mark DISCONT, we did a seek to another position");
for (walk = src->streams; walk; walk = g_list_next (walk)) {
}
break;
}
+ case GST_QUERY_URI:
+ {
+ gchar *uri;
+
+ uri = gst_rtspsrc_uri_get_uri (GST_URI_HANDLER (src));
+ if (uri != NULL) {
+ gst_query_set_uri (query, uri);
+ g_free (uri);
+ res = TRUE;
+ }
+ break;
+ }
default:
{
GstPad *target = gst_ghost_pad_get_target (GST_GHOST_PAD_CAST (pad));
gchar *name;
GstPadTemplate *template;
gint id, ssrc, pt;
- GList *lstream;
+ GList *ostreams;
GstRTSPStream *stream;
gboolean all_added;
if (sscanf (name, "recv_rtp_src_%u_%u_%u", &id, &ssrc, &pt) != 3)
goto unknown_stream;
- GST_DEBUG_OBJECT (src, "stream: %u, SSRC %d, PT %d", id, ssrc, pt);
+ GST_DEBUG_OBJECT (src, "stream: %u, SSRC %08x, PT %d", id, ssrc, pt);
stream = find_stream (src, &id, (gpointer) find_stream_by_id);
if (stream == NULL)
goto unknown_stream;
- /* create a new pad we will use to stream to */
- template = gst_static_pad_template_get (&rtptemplate);
- stream->srcpad = gst_ghost_pad_new_from_template (name, pad, template);
- gst_object_unref (template);
- g_free (name);
+ /* save SSRC */
+ stream->ssrc = ssrc;
+ /* we'll add it later see below */
stream->added = TRUE;
- gst_pad_set_event_function (stream->srcpad, gst_rtspsrc_handle_src_event);
- gst_pad_set_query_function (stream->srcpad, gst_rtspsrc_handle_src_query);
- gst_pad_set_active (stream->srcpad, TRUE);
- gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
/* check if we added all streams */
all_added = TRUE;
- for (lstream = src->streams; lstream; lstream = g_list_next (lstream)) {
- stream = (GstRTSPStream *) lstream->data;
+ for (ostreams = src->streams; ostreams; ostreams = g_list_next (ostreams)) {
+ GstRTSPStream *ostream = (GstRTSPStream *) ostreams->data;
GST_DEBUG_OBJECT (src, "stream %p, container %d, disabled %d, added %d",
- stream, stream->container, stream->disabled, stream->added);
+ ostream, ostream->container, ostream->disabled, ostream->added);
/* a container stream only needs one pad added. Also disabled streams don't
* count */
- if (!stream->container && !stream->disabled && !stream->added) {
+ if (!ostream->container && !ostream->disabled && !ostream->added) {
all_added = FALSE;
break;
}
}
GST_RTSP_STATE_UNLOCK (src);
+ /* create a new pad we will use to stream to */
+ template = gst_static_pad_template_get (&rtptemplate);
+ stream->srcpad = gst_ghost_pad_new_from_template (name, pad, template);
+ gst_object_unref (template);
+ g_free (name);
+
+ gst_pad_set_event_function (stream->srcpad, gst_rtspsrc_handle_src_event);
+ gst_pad_set_query_function (stream->srcpad, gst_rtspsrc_handle_src_query);
+ gst_pad_set_active (stream->srcpad, TRUE);
+ gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
+
if (all_added) {
GST_DEBUG_OBJECT (src, "We added all streams");
/* when we get here, all stream are added and we can fire the no-more-pads
on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
{
GstRTSPSrc *src = stream->parent;
+ guint ssrc;
+
+ g_object_get (source, "ssrc", &ssrc, NULL);
- GST_DEBUG_OBJECT (src, "source in session %u received BYE", stream->id);
+ GST_DEBUG_OBJECT (src, "source %08x, stream %08x, session %u received BYE",
+ ssrc, stream->ssrc, stream->id);
- gst_rtspsrc_do_stream_eos (src, stream);
+ if (ssrc == stream->ssrc)
+ gst_rtspsrc_do_stream_eos (src, stream);
}
static void
on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
{
GstRTSPSrc *src = stream->parent;
+ guint ssrc;
- GST_DEBUG_OBJECT (src, "source in session %u timed out", stream->id);
+ g_object_get (source, "ssrc", &ssrc, NULL);
- gst_rtspsrc_do_stream_eos (src, stream);
+ GST_WARNING_OBJECT (src, "source %08x, stream %08x in session %u timed out",
+ ssrc, stream->ssrc, stream->id);
+
+ if (ssrc == stream->ssrc)
+ gst_rtspsrc_do_stream_eos (src, stream);
}
static void
/* configure the manager */
if (src->manager == NULL) {
GObjectClass *klass;
- GstState target;
- if (!(src->manager = gst_element_factory_make (manager, NULL))) {
+ if (!(src->manager = gst_element_factory_make (manager, "manager"))) {
/* fallback */
if (gst_rtsp_transport_get_manager (transport->trans, &manager, 1) < 0)
goto no_manager;
if (!manager)
goto use_no_manager;
- if (!(src->manager = gst_element_factory_make (manager, NULL)))
+ if (!(src->manager = gst_element_factory_make (manager, "manager")))
goto manager_failed;
}
/* we manage this element */
+ gst_element_set_locked_state (src->manager, TRUE);
gst_bin_add (GST_BIN_CAST (src), src->manager);
- GST_OBJECT_LOCK (src);
- target = GST_STATE_TARGET (src);
- GST_OBJECT_UNLOCK (src);
-
- ret = gst_element_set_state (src->manager, target);
+ ret = gst_element_set_state (src->manager, GST_STATE_PAUSED);
if (ret == GST_STATE_CHANGE_FAILURE)
goto start_manager_failure;
g_object_set (src->manager, "latency", src->latency, NULL);
klass = G_OBJECT_GET_CLASS (G_OBJECT (src->manager));
+
+ if (g_object_class_find_property (klass, "ntp-sync")) {
+ g_object_set (src->manager, "ntp-sync", src->ntp_sync, NULL);
+ }
+
+ if (g_object_class_find_property (klass, "use-pipeline-clock")) {
+ g_object_set (src->manager, "use-pipeline-clock",
+ src->use_pipeline_clock, NULL);
+ }
+
+ if (g_object_class_find_property (klass, "drop-on-latency")) {
+ g_object_set (src->manager, "drop-on-latency", src->drop_on_latency,
+ NULL);
+ }
+
if (g_object_class_find_property (klass, "buffer-mode")) {
if (src->buffer_mode != BUFFER_MODE_AUTO) {
g_object_set (src->manager, "buffer-mode", src->buffer_mode, NULL);
g_object_set (rtpsession, "rtcp-rs-bandwidth", stream->rs_bandwidth,
NULL);
}
+
+ g_object_set (rtpsession, "probation", src->probation, NULL);
+
g_signal_connect (rtpsession, "on-bye-ssrc", (GCallback) on_bye_ssrc,
stream);
g_signal_connect (rtpsession, "on-bye-timeout", (GCallback) on_timeout,
for (i = 0; i < 2; i++) {
if (stream->udpsrc[i]) {
+ GST_DEBUG ("free UDP source %d for stream %p", i, stream);
gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL);
gst_object_unref (stream->udpsrc[i]);
stream->udpsrc[i] = NULL;
g_object_set (G_OBJECT (stream->udpsrc[0]), "buffer-size",
src->udp_buffer_size, NULL);
+ if (src->multi_iface != NULL)
+ g_object_set (G_OBJECT (stream->udpsrc[0]), "multicast-iface",
+ src->multi_iface, NULL);
+
/* change state */
+ gst_element_set_locked_state (stream->udpsrc[0], TRUE);
gst_element_set_state (stream->udpsrc[0], GST_STATE_PAUSED);
}
/* creating another UDP source for RTCP */
if (max != -1) {
+ GstCaps *caps;
+
uri = g_strdup_printf ("udp://%s:%d", destination, max);
stream->udpsrc[1] =
gst_element_make_from_uri (GST_URI_SRC, uri, NULL, NULL);
if (stream->udpsrc[1] == NULL)
goto no_element;
+ caps = gst_caps_new_empty_simple ("application/x-rtcp");
+ g_object_set (stream->udpsrc[1], "caps", caps, NULL);
+ gst_caps_unref (caps);
+
/* take ownership */
gst_object_ref_sink (stream->udpsrc[1]);
+ if (src->multi_iface != NULL)
+ g_object_set (G_OBJECT (stream->udpsrc[0]), "multicast-iface",
+ src->multi_iface, NULL);
+
gst_element_set_state (stream->udpsrc[1], GST_STATE_PAUSED);
}
return TRUE;
/* we manage the UDP elements now. For unicast, the UDP sources where
* allocated in the stream when we suggested a transport. */
if (stream->udpsrc[0]) {
+ gst_element_set_locked_state (stream->udpsrc[0], TRUE);
gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[0]);
GST_DEBUG_OBJECT (src, "setting up UDP source");
/* configure a timeout on the UDP port. When the timeout message is
* posted, we assume UDP transport is not possible. We reconnect using TCP
* if we can. */
- g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout", src->udp_timeout,
- NULL);
+ g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout",
+ src->udp_timeout * 1000, NULL);
/* get output pad of the UDP source. */
*outpad = gst_element_get_static_pad (stream->udpsrc[0], "src");
/* RTCP port */
if (stream->udpsrc[1]) {
+ GstCaps *caps;
+
+ gst_element_set_locked_state (stream->udpsrc[1], TRUE);
gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[1]);
+ caps = gst_caps_new_empty_simple ("application/x-rtcp");
+ g_object_set (stream->udpsrc[1], "caps", caps, NULL);
+ gst_caps_unref (caps);
+
if (stream->channelpad[1]) {
GstPad *pad;
gboolean res = TRUE;
/* only streams that have a connection to the outside world */
- if (stream->srcpad == NULL)
+ if (stream->container || stream->disabled)
goto done;
if (stream->udpsrc[0]) {
GST_DEBUG_OBJECT (src, "set flushing %d", flush);
GST_RTSP_STATE_LOCK (src);
- if (src->conninfo.connection) {
+ if (src->conninfo.connection && src->conninfo.flushing != flush) {
GST_DEBUG_OBJECT (src, "connection flush");
gst_rtsp_connection_flush (src->conninfo.connection, flush);
+ src->conninfo.flushing = flush;
}
for (walk = src->streams; walk; walk = g_list_next (walk)) {
GstRTSPStream *stream = (GstRTSPStream *) walk->data;
- GST_DEBUG_OBJECT (src, "stream %p flush", stream);
- if (stream->conninfo.connection)
+ if (stream->conninfo.connection && stream->conninfo.flushing != flush) {
+ GST_DEBUG_OBJECT (src, "stream %p flush", stream);
gst_rtsp_connection_flush (stream->conninfo.connection, flush);
+ stream->conninfo.flushing = flush;
+ }
}
GST_RTSP_STATE_UNLOCK (src);
}
if (res == GST_RTSP_ENOTIMPL) {
/* default implementation, send OK */
+ GST_DEBUG_OBJECT (src, "prepare OK reply");
res =
gst_rtsp_message_init_response (&response, GST_RTSP_STS_OK, "OK",
request);
if (res < 0)
goto send_error;
- GST_DEBUG_OBJECT (src, "replying with OK");
+ /* let app parse and reply */
+ g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_HANDLE_REQUEST],
+ 0, request, response);
if (src->debug)
gst_rtsp_message_dump (&response);
}
static GstFlowReturn
-gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
+gst_rtspsrc_handle_data (GstRTSPSrc * src, GstRTSPMessage * message)
{
- GstRTSPMessage message = { 0 };
- GstRTSPResult res;
+ GstFlowReturn ret = GST_FLOW_OK;
gint channel;
GstRTSPStream *stream;
GstPad *outpad = NULL;
guint8 *data;
guint size;
- GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *buf;
- gboolean is_rtcp, have_data;
-
- /* here we are only interested in data messages */
- have_data = FALSE;
- do {
- GTimeVal tv_timeout;
-
- /* get the next timeout interval */
- gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
-
- /* see if the timeout period expired */
- if ((tv_timeout.tv_sec | tv_timeout.tv_usec) == 0) {
- GST_DEBUG_OBJECT (src, "timout, sending keep-alive");
- /* send keep-alive, only act on interrupt, a warning will be posted for
- * other errors. */
- if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
- goto interrupt;
- /* get new timeout */
- gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
- }
-
- GST_DEBUG_OBJECT (src, "doing receive with timeout %ld seconds, %ld usec",
- tv_timeout.tv_sec, tv_timeout.tv_usec);
-
- /* protect the connection with the connection lock so that we can see when
- * we are finished doing server communication */
- res =
- gst_rtspsrc_connection_receive (src, src->conninfo.connection,
- &message, src->ptcp_timeout);
-
- switch (res) {
- case GST_RTSP_OK:
- GST_DEBUG_OBJECT (src, "we received a server message");
- break;
- case GST_RTSP_EINTR:
- /* we got interrupted this means we need to stop */
- goto interrupt;
- case GST_RTSP_ETIMEOUT:
- /* no reply, send keep alive */
- GST_DEBUG_OBJECT (src, "timeout, sending keep-alive");
- if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
- goto interrupt;
- continue;
- case GST_RTSP_EEOF:
- /* go EOS when the server closed the connection */
- goto server_eof;
- default:
- goto receive_error;
- }
-
- switch (message.type) {
- case GST_RTSP_MESSAGE_REQUEST:
- /* server sends us a request message, handle it */
- res =
- gst_rtspsrc_handle_request (src, src->conninfo.connection,
- &message);
- if (res == GST_RTSP_EEOF)
- goto server_eof;
- else if (res < 0)
- goto handle_request_failed;
- break;
- case GST_RTSP_MESSAGE_RESPONSE:
- /* we ignore response messages */
- GST_DEBUG_OBJECT (src, "ignoring response message");
- if (src->debug)
- gst_rtsp_message_dump (&message);
- break;
- case GST_RTSP_MESSAGE_DATA:
- GST_DEBUG_OBJECT (src, "got data message");
- have_data = TRUE;
- break;
- default:
- GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
- message.type);
- break;
- }
- }
- while (!have_data);
+ gboolean is_rtcp;
+ GstEvent *event;
- channel = message.type_data.data.channel;
+ channel = message->type_data.data.channel;
stream = find_stream (src, &channel, (gpointer) find_stream_by_channel);
if (!stream)
}
/* take a look at the body to figure out what we have */
- gst_rtsp_message_get_body (&message, &data, &size);
+ gst_rtsp_message_get_body (message, &data, &size);
if (size < 2)
goto invalid_length;
goto unknown_stream;
/* take the message body for further processing */
- gst_rtsp_message_steal_body (&message, &data, &size);
+ gst_rtsp_message_steal_body (message, &data, &size);
/* strip the trailing \0 */
size -= 1;
gst_memory_new_wrapped (0, data, size, 0, size, data, g_free));
/* don't need message anymore */
- gst_rtsp_message_unset (&message);
+ gst_rtsp_message_unset (message);
GST_DEBUG_OBJECT (src, "pushing data of size %d on channel %d", size,
channel);
if (src->need_activate) {
+ gchar *stream_id;
+ GstEvent *event;
+ GChecksum *cs;
+ gchar *uri;
+
+ /* generate an SHA256 sum of the URI */
+ cs = g_checksum_new (G_CHECKSUM_SHA256);
+ uri = src->conninfo.location;
+ g_checksum_update (cs, (const guchar *) uri, strlen (uri));
+ stream_id =
+ g_strdup_printf ("%s/%d", g_checksum_get_string (cs), stream->id);
+ g_checksum_free (cs);
+ event = gst_event_new_stream_start (stream_id);
+ g_free (stream_id);
+ gst_rtspsrc_push_event (src, event);
+
gst_rtspsrc_activate_streams (src);
src->need_activate = FALSE;
}
+ if ((event = src->start_segment) != NULL) {
+ src->start_segment = NULL;
+ gst_rtspsrc_push_event (src, event);
+ }
if (src->base_time == -1) {
/* Take current running_time. This timestamp will be put on
unknown_stream:
{
GST_DEBUG_OBJECT (src, "unknown stream on channel %d, ignored", channel);
- gst_rtsp_message_unset (&message);
+ gst_rtsp_message_unset (message);
+ return GST_FLOW_OK;
+ }
+invalid_length:
+ {
+ GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
+ ("Short message received, ignoring."));
+ gst_rtsp_message_unset (message);
return GST_FLOW_OK;
}
+}
+
+static GstFlowReturn
+gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
+{
+ GstRTSPMessage message = { 0 };
+ GstRTSPResult res;
+ GstFlowReturn ret = GST_FLOW_OK;
+ GTimeVal tv_timeout;
+
+ while (TRUE) {
+ /* get the next timeout interval */
+ gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
+
+ /* see if the timeout period expired */
+ if ((tv_timeout.tv_sec | tv_timeout.tv_usec) == 0) {
+ GST_DEBUG_OBJECT (src, "timout, sending keep-alive");
+ /* send keep-alive, only act on interrupt, a warning will be posted for
+ * other errors. */
+ if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
+ goto interrupt;
+ /* get new timeout */
+ gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
+ }
+
+ GST_DEBUG_OBJECT (src, "doing receive with timeout %ld seconds, %ld usec",
+ tv_timeout.tv_sec, tv_timeout.tv_usec);
+
+ /* protect the connection with the connection lock so that we can see when
+ * we are finished doing server communication */
+ res =
+ gst_rtspsrc_connection_receive (src, src->conninfo.connection,
+ &message, src->ptcp_timeout);
+
+ switch (res) {
+ case GST_RTSP_OK:
+ GST_DEBUG_OBJECT (src, "we received a server message");
+ break;
+ case GST_RTSP_EINTR:
+ /* we got interrupted this means we need to stop */
+ goto interrupt;
+ case GST_RTSP_ETIMEOUT:
+ /* no reply, send keep alive */
+ GST_DEBUG_OBJECT (src, "timeout, sending keep-alive");
+ if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
+ goto interrupt;
+ continue;
+ case GST_RTSP_EEOF:
+ /* go EOS when the server closed the connection */
+ goto server_eof;
+ default:
+ goto receive_error;
+ }
+
+ switch (message.type) {
+ case GST_RTSP_MESSAGE_REQUEST:
+ /* server sends us a request message, handle it */
+ res =
+ gst_rtspsrc_handle_request (src, src->conninfo.connection,
+ &message);
+ if (res == GST_RTSP_EEOF)
+ goto server_eof;
+ else if (res < 0)
+ goto handle_request_failed;
+ break;
+ case GST_RTSP_MESSAGE_RESPONSE:
+ /* we ignore response messages */
+ GST_DEBUG_OBJECT (src, "ignoring response message");
+ if (src->debug)
+ gst_rtsp_message_dump (&message);
+ break;
+ case GST_RTSP_MESSAGE_DATA:
+ GST_DEBUG_OBJECT (src, "got data message");
+ ret = gst_rtspsrc_handle_data (src, &message);
+ if (ret != GST_FLOW_OK)
+ goto handle_data_failed;
+ break;
+ default:
+ GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
+ message.type);
+ break;
+ }
+ }
+ g_assert_not_reached ();
+
+ /* ERRORS */
server_eof:
{
GST_DEBUG_OBJECT (src, "we got an eof from the server");
interrupt:
{
gst_rtsp_message_unset (&message);
- GST_DEBUG_OBJECT (src, "got interrupted: stop connection flush");
- gst_rtspsrc_connection_flush (src, FALSE);
+ GST_DEBUG_OBJECT (src, "got interrupted");
return GST_FLOW_FLUSHING;
}
receive_error:
gst_rtsp_message_unset (&message);
return GST_FLOW_ERROR;
}
-invalid_length:
+handle_data_failed:
{
- GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
- ("Short message received, ignoring."));
- gst_rtsp_message_unset (&message);
- return GST_FLOW_OK;
+ GST_DEBUG_OBJECT (src, "could no handle data message");
+ return ret;
}
}
* see what happens. */
GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
("The server closed the connection."));
- if ((res =
- gst_rtsp_conninfo_reconnect (src, &src->conninfo, FALSE)) < 0)
- goto connect_error;
-
+ if (src->udp_reconnect) {
+ if ((res =
+ gst_rtsp_conninfo_reconnect (src, &src->conninfo, FALSE)) < 0)
+ goto connect_error;
+ } else {
+ goto server_eof;
+ }
continue;
case GST_RTSP_ENET:
GST_DEBUG_OBJECT (src, "An ethernet problem occured.");
break;
}
}
+ g_assert_not_reached ();
/* we get here when the connection got interrupted */
interrupt:
{
gst_rtsp_message_unset (&message);
- GST_DEBUG_OBJECT (src, "got interrupted: stop connection flush");
- gst_rtspsrc_connection_flush (src, FALSE);
+ GST_DEBUG_OBJECT (src, "got interrupted");
return GST_FLOW_FLUSHING;
}
connect_error:
GST_OBJECT_LOCK (src);
old = src->pending_cmd;
+ if (old == CMD_RECONNECT) {
+ GST_DEBUG_OBJECT (src, "ignore, we were reconnecting");
+ cmd = CMD_RECONNECT;
+ }
if (old != CMD_WAIT) {
src->pending_cmd = CMD_WAIT;
GST_OBJECT_UNLOCK (src);
/* cancel previous request */
+ GST_DEBUG_OBJECT (src, "cancel previous request %d", old);
gst_rtspsrc_loop_cancel_cmd (src, old);
GST_OBJECT_LOCK (src);
}
("streaming task paused, reason %s (%d)", reason, ret));
gst_rtspsrc_push_event (src, gst_event_new_eos ());
}
+ gst_rtspsrc_loop_send_cmd (src, CMD_WAIT, CMD_LOOP);
return FALSE;
}
}
break;
case GST_RTSP_MESSAGE_DATA:
/* get next response */
- GST_DEBUG_OBJECT (src, "ignoring data response message");
+ GST_DEBUG_OBJECT (src, "handle data response message");
+ gst_rtspsrc_handle_data (src, response);
goto next;
default:
GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
{
switch (res) {
case GST_RTSP_EEOF:
- GST_WARNING_OBJECT (src, "server closed connection, doing reconnect");
- if (try == 0) {
+ GST_WARNING_OBJECT (src, "server closed connection");
+ if ((try == 0) && !src->interleaved && src->udp_reconnect) {
try++;
/* if reconnect succeeds, try again */
if ((res =
{
GstRTSPHeaderField field;
gchar *respoptions;
- gchar **options;
gint indx = 0;
- gint i;
/* reset supported methods */
src->methods = 0;
if (!respoptions)
break;
- /* If we get here, the server gave a list of supported methods, parse
- * them here. The string is like:
- *
- * OPTIONS, DESCRIBE, ANNOUNCE, PLAY, SETUP, ...
- */
- options = g_strsplit (respoptions, ",", 0);
-
- for (i = 0; options[i]; i++) {
- gchar *stripped;
- gint method;
-
- stripped = g_strstrip (options[i]);
- method = gst_rtsp_find_method (stripped);
-
- /* keep bitfield of supported methods */
- if (method != GST_RTSP_INVALID)
- src->methods |= method;
- }
- g_strfreev (options);
+ src->methods |= gst_rtsp_options_from_text (respoptions);
indx++;
}
if (add_udp_str)
g_string_append (result, "/UDP");
g_string_append (result, ";multicast");
+ if (src->next_port_num != 0) {
+ if (src->client_port_range.max > 0 &&
+ src->next_port_num >= src->client_port_range.max)
+ goto no_ports;
+
+ g_string_append_printf (result, ";client_port=%d-%d",
+ src->next_port_num, src->next_port_num + 1);
+ }
} else if (protocols & GST_RTSP_LOWER_TRANS_TCP) {
GST_DEBUG_OBJECT (src, "adding TCP");
/* ERRORS */
failed:
{
+ GST_ERROR ("extension gave error %d", res);
return res;
}
+no_ports:
+ {
+ GST_ERROR ("no more ports available");
+ return GST_RTSP_ERROR;
+ }
}
static GstRTSPResult
/* ERRORS */
failed:
{
+ GST_ERROR ("failed to allocate udp ports");
return GST_RTSP_ERROR;
}
}
/* only allow multicast for other streams */
GST_DEBUG_OBJECT (src, "stream %p as UDP multicast", stream);
protocols = GST_RTSP_LOWER_TRANS_UDP_MCAST;
+ /* if the server selected our ports, increment our counters so that
+ * we select a new port later */
+ if (src->next_port_num == transport.port.min &&
+ src->next_port_num + 1 == transport.port.max) {
+ src->next_port_num += 2;
+ }
break;
case GST_RTSP_LOWER_TRANS_UDP:
/* only allow unicast for other streams */
return TRUE;
}
+/* Parse clock profived by the server with following syntax:
+ *
+ * "GstNetTimeProvider <wrapped-clock> <server-IP:port> <clock-time>"
+ */
+static gboolean
+gst_rtspsrc_parse_gst_clock (GstRTSPSrc * src, const gchar * gstclock)
+{
+ gboolean res = FALSE;
+
+ if (g_str_has_prefix (gstclock, "GstNetTimeProvider ")) {
+ gchar **fields = NULL, **parts = NULL;
+ gchar *remote_ip, *str;
+ gint port;
+ GstClockTime base_time;
+ GstClock *netclock;
+
+ fields = g_strsplit (gstclock, " ", 0);
+
+ /* wrapped clock, not very interesting for now */
+ if (fields[1] == NULL)
+ goto cleanup;
+
+ /* remote IP address and port */
+ if ((str = fields[2]) == NULL)
+ goto cleanup;
+
+ parts = g_strsplit (str, ":", 0);
+
+ if ((remote_ip = parts[0]) == NULL)
+ goto cleanup;
+
+ if ((str = parts[1]) == NULL)
+ goto cleanup;
+
+ port = atoi (str);
+ if (port == 0)
+ goto cleanup;
+
+ /* base-time */
+ if ((str = fields[3]) == NULL)
+ goto cleanup;
+
+ base_time = g_ascii_strtoull (str, NULL, 10);
+
+ netclock =
+ gst_net_client_clock_new ((gchar *) "GstRTSPClock", remote_ip, port,
+ base_time);
+
+ if (src->provided_clock)
+ gst_object_unref (src->provided_clock);
+ src->provided_clock = netclock;
+
+ gst_element_post_message (GST_ELEMENT_CAST (src),
+ gst_message_new_clock_provide (GST_OBJECT_CAST (src),
+ src->provided_clock, TRUE));
+
+ res = TRUE;
+ cleanup:
+ g_strfreev (fields);
+ g_strfreev (parts);
+ }
+ return res;
+}
+
/* must be called with the RTSP state lock */
static GstRTSPResult
gst_rtspsrc_open_from_sdp (GstRTSPSrc * src, GstSDPMessage * sdp,
break;
}
}
+ /* parse clock information. This is GStreamer specific, a server can tell the
+ * client what clock it is using and wrap that in a network clock. The
+ * advantage of that is that we can slave to it. */
+ {
+ const gchar *gstclock;
+
+ for (i = 0;; i++) {
+ gstclock = gst_sdp_message_get_attribute_val_n (sdp, "x-gst-clock", i);
+ if (gstclock == NULL)
+ break;
+
+ /* parse the clock and expose it in the provide_clock method */
+ if (gst_rtspsrc_parse_gst_clock (src, gstclock))
+ break;
+ }
+ }
/* try to find a global control attribute. Note that a '*' means that we should
* do aggregate control with the current url (so we don't do anything and
* leave the current connection as is) */
setup_failed:
{
GST_ERROR_OBJECT (src, "setup failed");
+ gst_rtspsrc_cleanup (src);
return res;
}
}
GST_DEBUG_OBJECT (src, "TEARDOWN...");
+ gst_rtspsrc_set_state (src, GST_STATE_READY);
+
if (src->state < GST_RTSP_STATE_READY) {
GST_DEBUG_OBJECT (src, "not ready, doing cleanup");
goto close;
* udp sources */
gst_rtspsrc_send_dummy_packets (src);
- /* activate receive elements;
- * only in async case, since receive elements may not have been affected
- * by overall state change (e.g. not around yet),
- * do not mess with state in sync case (e.g. seeking) */
- if (async)
- gst_element_set_state (GST_ELEMENT_CAST (src), GST_STATE_PLAYING);
+ /* require new SR packets */
+ if (src->manager)
+ g_signal_emit_by_name (src->manager, "reset-sync", NULL);
+
+ gst_rtspsrc_set_state (src, GST_STATE_PLAYING);
/* construct a control url */
if (src->control)
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_RANGE, hval);
g_free (hval);
+
+ /* store the newsegment event so it can be sent from the streaming thread. */
+ if (src->start_segment)
+ gst_event_unref (src->start_segment);
+ src->start_segment = gst_event_new_segment (&src->segment);
}
if (segment->rate != 1.0) {
break;
}
+ /* change element states now */
+ gst_rtspsrc_set_state (src, GST_STATE_PAUSED);
+
no_connection:
src->state = GST_RTSP_STATE_READY;
GST_OBJECT_LOCK (src);
cmd = src->pending_cmd;
- if (cmd == CMD_PLAY || cmd == CMD_LOOP)
+ if (cmd == CMD_RECONNECT || cmd == CMD_PLAY || cmd == CMD_PAUSE
+ || cmd == CMD_LOOP)
src->pending_cmd = CMD_LOOP;
else
src->pending_cmd = CMD_WAIT;
GST_DEBUG_OBJECT (src, "stopping");
/* also cancels pending task */
- gst_rtspsrc_loop_send_cmd (src, CMD_WAIT, CMD_CLOSE);
+ gst_rtspsrc_loop_send_cmd (src, CMD_WAIT, CMD_ALL);
GST_OBJECT_LOCK (src);
if ((task = src->task)) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
/* unblock the tcp tasks and make the loop waiting */
gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_WAIT, CMD_LOOP);
+ /* make sure it is waiting before we send PAUSE or PLAY below */
+ GST_RTSP_STREAM_LOCK (rtspsrc);
+ GST_RTSP_STREAM_UNLOCK (rtspsrc);
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
break;
goto done;
switch (transition) {
+ case GST_STATE_CHANGE_NULL_TO_READY:
+ ret = GST_STATE_CHANGE_SUCCESS;
+ break;
+ case GST_STATE_CHANGE_READY_TO_PAUSED:
+ ret = GST_STATE_CHANGE_NO_PREROLL;
+ break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PLAY, 0);
+ ret = GST_STATE_CHANGE_SUCCESS;
break;
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
/* send pause request and keep the idle task around */
gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PAUSE, CMD_LOOP);
ret = GST_STATE_CHANGE_NO_PREROLL;
break;
- case GST_STATE_CHANGE_READY_TO_PAUSED:
- ret = GST_STATE_CHANGE_NO_PREROLL;
- break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_CLOSE, CMD_PAUSE);
+ ret = GST_STATE_CHANGE_SUCCESS;
break;
case GST_STATE_CHANGE_READY_TO_NULL:
gst_rtspsrc_stop (rtspsrc);
+ ret = GST_STATE_CHANGE_SUCCESS;
break;
default:
break;
gst_rtspsrc_uri_get_protocols (GType type)
{
static const gchar *protocols[] =
- { "rtsp", "rtspu", "rtspt", "rtsph", "rtsp-sdp", NULL };
+ { "rtsp", "rtspu", "rtspt", "rtsph", "rtsp-sdp",
+ "rtsps", "rtspsu", "rtspst", "rtspsh", NULL
+ };
return protocols;
}
projects / platform / upstream / gstreamer.git / blobdiff