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rtspsrc: improve RTP session handling

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Store the RTP session in the stream so that we can more efficiently
perform actions on the stream based on RTP signals.
This commit is contained in:
Wim Taymans 2010-12-23 15:24:29 +01:00
parent 7caad21a57
commit 12bc7258b9
2 changed files with 90 additions and 84 deletions
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165
gst/rtsp/gstrtspsrc.c
View file

@ -1126,6 +1126,10 @@ gst_rtspsrc_stream_free (GstRTSPSrc * src, GstRTSPStream * stream)
gst_object_unref (stream->rtcppad);
stream->rtcppad = NULL;
}
if (stream->session) {
g_object_unref (stream->session);
stream->session = NULL;
}
g_free (stream);
}
@ -1143,14 +1147,14 @@ gst_rtspsrc_cleanup (GstRTSPSrc * src)
}
g_list_free (src->streams);
src->streams = NULL;
if (src->session) {
if (src->session_sig_id) {
g_signal_handler_disconnect (src->session, src->session_sig_id);
src->session_sig_id = 0;
if (src->manager) {
if (src->manager_sig_id) {
g_signal_handler_disconnect (src->manager, src->manager_sig_id);
src->manager_sig_id = 0;
}
gst_element_set_state (src->session, GST_STATE_NULL);
gst_bin_remove (GST_BIN_CAST (src), src->session);
src->session = NULL;
gst_element_set_state (src->manager, GST_STATE_NULL);
gst_bin_remove (GST_BIN_CAST (src), src->manager);
src->manager = NULL;
}
src->numstreams = 0;
if (src->props)
@ -1672,8 +1676,8 @@ gst_rtspsrc_flush (GstRTSPSrc * src, gboolean flush)
if (base_time != -1)
gst_element_set_base_time (GST_ELEMENT_CAST (src), base_time);
/* to manage jitterbuffer buffer mode */
if (src->session)
gst_element_set_base_time (GST_ELEMENT_CAST (src->session), base_time);
if (src->manager)
gst_element_set_base_time (GST_ELEMENT_CAST (src->manager), base_time);
}
static GstRTSPResult
@ -2138,7 +2142,7 @@ was_ok:
/* this callback is called when the session manager generated a new src pad with
* payloaded RTP packets. We simply ghost the pad here. */
static void
new_session_pad (GstElement * session, GstPad * pad, GstRTSPSrc * src)
new_manager_pad (GstElement * manager, GstPad * pad, GstRTSPSrc * src)
{
gchar *name;
GstPadTemplate *template;
@ -2147,7 +2151,7 @@ new_session_pad (GstElement * session, GstPad * pad, GstRTSPSrc * src)
GstRTSPStream *stream;
gboolean all_added;
GST_DEBUG_OBJECT (src, "got new session pad %" GST_PTR_FORMAT, pad);
GST_DEBUG_OBJECT (src, "got new manager pad %" GST_PTR_FORMAT, pad);
GST_RTSP_STATE_LOCK (src);
/* find stream */
@ -2210,7 +2214,7 @@ unknown_stream:
}
static GstCaps *
request_pt_map (GstElement * sess, guint session, guint pt, GstRTSPSrc * src)
request_pt_map (GstElement * manager, guint session, guint pt, GstRTSPSrc * src)
{
GstRTSPStream *stream;
GstCaps *caps;
@ -2238,17 +2242,12 @@ unknown_stream:
}
static void
gst_rtspsrc_do_stream_eos (GstRTSPSrc * src, guint session)
gst_rtspsrc_do_stream_eos (GstRTSPSrc * src, GstRTSPStream * stream)
{
GstRTSPStream *stream;
guint session = stream->id;
GST_DEBUG_OBJECT (src, "setting stream for session %u to EOS", session);
/* get stream for session */
stream = find_stream (src, &session, (gpointer) find_stream_by_id);
if (!stream)
goto unknown_stream;
if (stream->eos)
goto was_eos;
@ -2257,11 +2256,6 @@ gst_rtspsrc_do_stream_eos (GstRTSPSrc * src, guint session)
return;
/* ERRORS */
unknown_stream:
{
GST_DEBUG_OBJECT (src, "unknown stream for session %u", session);
return;
}
was_eos:
{
GST_DEBUG_OBJECT (src, "stream for session %u was already EOS", session);
@ -2270,30 +2264,41 @@ was_eos:
}
static void
on_bye_ssrc (GstElement * manager, guint session, guint32 ssrc,
GstRTSPSrc * src)
on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
{
GST_DEBUG_OBJECT (src, "SSRC %08x in session %u received BYE", ssrc, session);
GstRTSPSrc *src = stream->parent;
gst_rtspsrc_do_stream_eos (src, session);
GST_DEBUG_OBJECT (src, "source in session %u received BYE", stream->id);
gst_rtspsrc_do_stream_eos (src, stream);
}
static void
on_timeout (GstElement * manager, guint session, guint32 ssrc, GstRTSPSrc * src)
on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
{
GST_DEBUG_OBJECT (src, "SSRC %08x in session %u timed out", ssrc, session);
GstRTSPSrc *src = stream->parent;
gst_rtspsrc_do_stream_eos (src, session);
GST_DEBUG_OBJECT (src, "source in session %u timed out", stream->id);
gst_rtspsrc_do_stream_eos (src, stream);
}
static void
on_npt_stop (GstElement * manager, guint session, guint32 ssrc,
GstRTSPSrc * src)
on_npt_stop (GObject * session, GObject * source, GstRTSPStream * stream)
{
GST_DEBUG_OBJECT (src, "SSRC %08x in session %u reached the NPT stop", ssrc,
session);
GstRTSPSrc *src = stream->parent;
gst_rtspsrc_do_stream_eos (src, session);
GST_DEBUG_OBJECT (src, "source in session %u reached NPT stop", stream->id);
gst_rtspsrc_do_stream_eos (src, stream);
}
static void
on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
{
GstRTSPSrc *src = stream->parent;
GST_DEBUG_OBJECT (src, "source in session %u is active", stream->id);
}
/* try to get and configure a manager */
@ -2313,11 +2318,11 @@ gst_rtspsrc_stream_configure_manager (GstRTSPSrc * src, GstRTSPStream * stream,
GST_DEBUG_OBJECT (src, "using manager %s", manager);
/* configure the manager */
if (src->session == NULL) {
if (src->manager == NULL) {
GObjectClass *klass;
GstState target;
if (!(src->session = gst_element_factory_make (manager, NULL))) {
if (!(src->manager = gst_element_factory_make (manager, NULL))) {
/* fallback */
if (gst_rtsp_transport_get_manager (transport->trans, &manager, 1) < 0)
goto no_manager;
@ -2325,27 +2330,27 @@ gst_rtspsrc_stream_configure_manager (GstRTSPSrc * src, GstRTSPStream * stream,
if (!manager)
goto use_no_manager;
if (!(src->session = gst_element_factory_make (manager, NULL)))
if (!(src->manager = gst_element_factory_make (manager, NULL)))
goto manager_failed;
}
/* we manage this element */
gst_bin_add (GST_BIN_CAST (src), src->session);
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->session, target);
ret = gst_element_set_state (src->manager, target);
if (ret == GST_STATE_CHANGE_FAILURE)
goto start_session_failure;
goto start_manager_failure;
g_object_set (src->session, "latency", src->latency, NULL);
g_object_set (src->manager, "latency", src->latency, NULL);
klass = G_OBJECT_GET_CLASS (G_OBJECT (src->session));
klass = G_OBJECT_GET_CLASS (G_OBJECT (src->manager));
if (g_object_class_find_property (klass, "buffer-mode")) {
if (src->buffer_mode != BUFFER_MODE_AUTO) {
g_object_set (src->session, "buffer-mode", src->buffer_mode, NULL);
g_object_set (src->manager, "buffer-mode", src->buffer_mode, NULL);
} else {
gboolean need_slave;
GstStructure *s;
@ -2368,11 +2373,11 @@ gst_rtspsrc_stream_configure_manager (GstRTSPSrc * src, GstRTSPStream * stream,
if (GST_CLOCK_TIME_IS_VALID (src->segment.duration) &&
src->segment.duration && !need_slave) {
GST_DEBUG_OBJECT (src, "selected buffer");
g_object_set (src->session, "buffer-mode", BUFFER_MODE_BUFFER,
g_object_set (src->manager, "buffer-mode", BUFFER_MODE_BUFFER,
NULL);
} else {
GST_DEBUG_OBJECT (src, "selected slave");
g_object_set (src->session, "buffer-mode", BUFFER_MODE_SLAVE, NULL);
g_object_set (src->manager, "buffer-mode", BUFFER_MODE_SLAVE, NULL);
}
}
}
@ -2380,46 +2385,35 @@ gst_rtspsrc_stream_configure_manager (GstRTSPSrc * src, GstRTSPStream * stream,
/* connect to signals if we did not already do so */
GST_DEBUG_OBJECT (src, "connect to signals on session manager, stream %p",
stream);
src->session_sig_id =
g_signal_connect (src->session, "pad-added",
(GCallback) new_session_pad, src);
src->session_ptmap_id =
g_signal_connect (src->session, "request-pt-map",
src->manager_sig_id =
g_signal_connect (src->manager, "pad-added",
(GCallback) new_manager_pad, src);
src->manager_ptmap_id =
g_signal_connect (src->manager, "request-pt-map",
(GCallback) request_pt_map, src);
g_signal_connect (src->session, "on-bye-ssrc", (GCallback) on_bye_ssrc,
src);
g_signal_connect (src->session, "on-bye-timeout", (GCallback) on_timeout,
src);
g_signal_connect (src->session, "on-timeout", (GCallback) on_timeout,
src);
/* FIXME: remove this once the rdtmanager is released */
if (g_signal_lookup ("on-npt-stop", G_OBJECT_TYPE (src->session)) != 0) {
g_signal_connect (src->session, "on-npt-stop", (GCallback) on_npt_stop,
src);
} else {
GST_INFO_OBJECT (src, "skipping on-npt-stop handling, not implemented");
}
}
/* we stream directly to the manager, get some pads. Each RTSP stream goes
* into a separate RTP session. */
name = g_strdup_printf ("recv_rtp_sink_%d", stream->id);
stream->channelpad[0] = gst_element_get_request_pad (src->session, name);
stream->channelpad[0] = gst_element_get_request_pad (src->manager, name);
g_free (name);
name = g_strdup_printf ("recv_rtcp_sink_%d", stream->id);
stream->channelpad[1] = gst_element_get_request_pad (src->session, name);
stream->channelpad[1] = gst_element_get_request_pad (src->manager, name);
g_free (name);
/* now configure the bandwidth in the session */
/* now configure the bandwidth in the manager */
if (g_signal_lookup ("get-internal-session",
G_OBJECT_TYPE (src->session)) != 0) {
G_OBJECT_TYPE (src->manager)) != 0) {
GObject *rtpsession;
g_signal_emit_by_name (src->session, "get-internal-session", stream->id,
g_signal_emit_by_name (src->manager, "get-internal-session", stream->id,
&rtpsession);
if (rtpsession) {
GST_INFO_OBJECT (src, "configure bandwidth in session %p", rtpsession);
stream->session = rtpsession;
if (stream->as_bandwidth != -1) {
GST_INFO_OBJECT (src, "setting AS: %f",
(gdouble) (stream->as_bandwidth * 1000));
@ -2436,7 +2430,16 @@ gst_rtspsrc_stream_configure_manager (GstRTSPSrc * src, GstRTSPStream * stream,
g_object_set (rtpsession, "rtcp-rs-bandwidth", stream->rs_bandwidth,
NULL);
}
g_object_unref (rtpsession);
g_signal_connect (rtpsession, "on-bye-ssrc", (GCallback) on_bye_ssrc,
stream);
g_signal_connect (rtpsession, "on-bye-timeout", (GCallback) on_timeout,
stream);
g_signal_connect (rtpsession, "on-timeout", (GCallback) on_timeout,
stream);
g_signal_connect (rtpsession, "on-npt-stop", (GCallback) on_npt_stop,
stream);
g_signal_connect (rtpsession, "on-ssrc-active",
(GCallback) on_ssrc_active, stream);
}
}
}
@ -2455,9 +2458,9 @@ manager_failed:
GST_DEBUG_OBJECT (src, "no session manager element %s found", manager);
return FALSE;
}
start_session_failure:
start_manager_failure:
{
GST_DEBUG_OBJECT (src, "could not start session");
GST_DEBUG_OBJECT (src, "could not start session manager");
return FALSE;
}
}
@ -2545,7 +2548,7 @@ gst_rtspsrc_stream_configure_tcp (GstRTSPSrc * src, GstRTSPStream * stream,
gst_object_unref (template);
}
/* setup RTCP transport back to the server if we have to. */
if (src->session && src->do_rtcp) {
if (src->manager && src->do_rtcp) {
GstPad *pad;
template = gst_static_pad_template_get (&anysinktemplate);
@ -2557,7 +2560,7 @@ gst_rtspsrc_stream_configure_tcp (GstRTSPSrc * src, GstRTSPStream * stream,
/* get session RTCP pad */
name = g_strdup_printf ("send_rtcp_src_%d", stream->id);
pad = gst_element_get_request_pad (src->session, name);
pad = gst_element_get_request_pad (src->manager, name);
g_free (name);
/* and link */
@ -2780,7 +2783,7 @@ gst_rtspsrc_stream_configure_udp_sinks (GstRTSPSrc * src,
do_rtp = (rtp_port != -1);
/* it's possible that the server does not want us to send RTCP in which case
* the port is -1 */
do_rtcp = (rtcp_port != -1 && src->session != NULL && src->do_rtcp);
do_rtcp = (rtcp_port != -1 && src->manager != NULL && src->do_rtcp);
/* we need a destination when we have RTP or RTCP ports */
if (destination == NULL && (do_rtp || do_rtcp))
@ -2889,7 +2892,7 @@ gst_rtspsrc_stream_configure_udp_sinks (GstRTSPSrc * src,
/* get session RTCP pad */
name = g_strdup_printf ("send_rtcp_src_%d", stream->id);
pad = gst_element_get_request_pad (src->session, name);
pad = gst_element_get_request_pad (src->manager, name);
g_free (name);
/* and link */
@ -3066,7 +3069,7 @@ gst_rtspsrc_activate_streams (GstRTSPSrc * src)
if (stream->srcpad) {
/* if we don't have a session manager, set the caps now. If we have a
* session, we will get a notification of the pad and the caps. */
if (!src->session) {
if (!src->manager) {
GST_DEBUG_OBJECT (src, "setting pad caps for stream %p", stream);
gst_pad_set_caps (stream->srcpad, stream->caps);
}
@ -3134,9 +3137,9 @@ gst_rtspsrc_configure_caps (GstRTSPSrc * src, GstSegment * segment)
}
GST_DEBUG_OBJECT (src, "stream %p, caps %" GST_PTR_FORMAT, stream, caps);
}
if (src->session) {
if (src->manager) {
GST_DEBUG_OBJECT (src, "clear session");
g_signal_emit_by_name (src->session, "clear-pt-map", NULL);
g_signal_emit_by_name (src->manager, "clear-pt-map", NULL);
}
}
@ -3592,7 +3595,7 @@ gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
src->need_activate = FALSE;
}
if (!src->session) {
if (!src->manager) {
/* set stream caps on buffer when we don't have a session manager to do it
* for us */
gst_buffer_set_caps (buf, stream->caps);

9
gst/rtsp/gstrtspsrc.h
View file

@ -138,6 +138,9 @@ struct _GstRTSPStream {
/* per stream connection */
GstRTSPConnInfo conninfo;
/* session */
GObject *session;
/* bandwidth */
guint as_bandwidth;
guint rs_bandwidth;
@ -233,9 +236,9 @@ struct _GstRTSPSrc {
gboolean seekable;
/* session management */
GstElement *session;
gulong session_sig_id;
gulong session_ptmap_id;
GstElement *manager;
gulong manager_sig_id;
gulong manager_ptmap_id;
GstRTSPConnInfo conninfo;