mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-07-08 15:45:55 +00:00
Code Issues 1 Projects Releases Wiki Activity

rtsp-media: Wait on async when needed.

Browse source 
Wait on asyn-done when needed in gst_rtsp_media_seek_trickmode.

In the unit test the pause from adjust_play_mode will cause a preroll
and after that async-done will be produced.
Without this patch there are no one consuming this async-done and when
later when seek fluch is done in gst_rtsp_media_seek_trickmode then it
wait for async-done. But then it wrongly find the async-done prodused by
adjus_play_mode and continue executing without waiting for the preroll
to finish.
This commit is contained in:
Göran Jönsson 2019-09-11 07:08:37 +02:00 committed by Sebastian Dröge
parent 7e1edcf1a4
commit 18f4f4e509
2 changed files with 169 additions and 4 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

47
gst/rtsp-server/rtsp-media.c
View file

@ -152,6 +152,7 @@ struct _GstRTSPMediaPrivate
/* Dynamic element handling */
guint nb_dynamic_elements;
guint no_more_pads_pending;
gboolean expected_async_done;
};
#define DEFAULT_SHARED FALSE
@ -476,6 +477,7 @@ gst_rtsp_media_init (GstRTSPMedia * media)
priv->max_mcast_ttl = DEFAULT_MAX_MCAST_TTL;
priv->bind_mcast_address = DEFAULT_BIND_MCAST_ADDRESS;
priv->do_rate_control = DEFAULT_DO_RATE_CONTROL;
priv->expected_async_done = FALSE;
}
static void
@ -2819,6 +2821,30 @@ gst_rtsp_media_seek_trickmode (GstRTSPMedia * media,
GstEvent *seek_event;
gboolean unblock = FALSE;
/* Handle expected async-done before waiting on next async-done.
*
* Since the seek further down in code will cause a preroll and
* a async-done will be generated it's important to wait on async-done
* if that is expected. Otherwise there is the risk that the waiting
* for async-done after the seek is detecting the expected async-done
* instead of the one that corresponds to the seek. Then execution
* continue and act as if the pipeline is prerolled, but it's not.
*
* During wait_preroll message GST_MESSAGE_ASYNC_DONE will come
* and then the state will change from preparing to prepared */
if (priv->expected_async_done) {
GST_DEBUG (" expected to get async-done, waiting ");
gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
g_rec_mutex_unlock (&priv->state_lock);
/* wait until pipeline is prerolled */
if (!wait_preroll (media))
goto preroll_failed_expected_async_done;
g_rec_mutex_lock (&priv->state_lock);
GST_DEBUG (" got expected async-done");
}
gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
if (rate < 0.0) {
@ -2910,6 +2936,11 @@ preroll_failed:
GST_WARNING ("failed to preroll after seek");
return FALSE;
}
preroll_failed_expected_async_done:
{
GST_WARNING ("failed to preroll");
return FALSE;
}
}
/**
@ -3118,6 +3149,8 @@ default_handle_message (GstRTSPMedia * media, GstMessage * message)
case GST_MESSAGE_STREAM_STATUS:
break;
case GST_MESSAGE_ASYNC_DONE:
if (priv->expected_async_done)
priv->expected_async_done = FALSE;
if (priv->complete) {
/* receive the final ASYNC_DONE, that is posted by the media pipeline
* after all the transport parts have been successfully added to
@ -4459,6 +4492,8 @@ static void
media_set_pipeline_state_locked (GstRTSPMedia * media, GstState state)
{
GstRTSPMediaPrivate *priv = media->priv;
GstStateChangeReturn set_state_ret;
priv->expected_async_done = FALSE;
if (state == GST_STATE_NULL) {
gst_rtsp_media_unprepare (media);
@ -4474,11 +4509,15 @@ media_set_pipeline_state_locked (GstRTSPMedia * media, GstState state)
/* make sure pads are not blocking anymore when going to PLAYING */
media_unblock_linked (media);
set_state (media, state);
/* and suspend after pause */
if (state == GST_STATE_PAUSED)
if (state == GST_STATE_PAUSED) {
set_state_ret = set_state (media, state);
if (set_state_ret == GST_STATE_CHANGE_ASYNC)
priv->expected_async_done = TRUE;
/* and suspend after pause */
gst_rtsp_media_suspend (media);
} else {
set_state (media, state);
}
}
}
}

126
tests/check/gst/rtspserver.c
View file

@ -483,6 +483,16 @@ do_simple_request (GstRTSPConnection * conn, GstRTSPMethod method,
NULL, NULL, NULL, NULL, NULL);
}
/* send an rtsp request with a method,session and range in,
* and receive response. range_in is the Range in req header */
static GstRTSPStatusCode
do_simple_request_rangein (GstRTSPConnection * conn, GstRTSPMethod method,
const gchar * session, const gchar * rangein)
{
return do_request (conn, method, NULL, session, NULL, rangein, NULL,
NULL, NULL, NULL, NULL, NULL);
}
/* send a DESCRIBE request and receive response. returns a received
* GstSDPMessage that must be freed by the caller */
static GstSDPMessage *
@ -2473,6 +2483,121 @@ GST_START_TEST (test_suspend_mode_reset_only_audio)
GST_END_TEST;
static GstRTSPStatusCode
adjust_play_mode (GstRTSPClient * client, GstRTSPContext * ctx,
GstRTSPTimeRange ** range, GstSeekFlags * flags, gdouble * rate,
GstClockTime * trickmode_interval, gboolean * enable_rate_control)
{
GstRTSPState rtspstate;
rtspstate = gst_rtsp_session_media_get_rtsp_state (ctx->sessmedia);
if (rtspstate == GST_RTSP_STATE_PLAYING) {
if (!gst_rtsp_session_media_set_state (ctx->sessmedia, GST_STATE_PAUSED))
return GST_RTSP_STS_INTERNAL_SERVER_ERROR;
if (!gst_rtsp_media_unsuspend (ctx->media))
return GST_RTSP_STS_INTERNAL_SERVER_ERROR;
}
return GST_RTSP_STS_OK;
}
GST_START_TEST (test_double_play)
{
GstRTSPMountPoints *mounts;
gchar *service;
GstRTSPMediaFactory *factory;
GstRTSPConnection *conn;
GstSDPMessage *sdp_message = NULL;
const GstSDPMedia *sdp_media;
const gchar *video_control;
const gchar *audio_control;
GstRTSPRange client_port;
gchar *session = NULL;
GstRTSPTransport *audio_transport = NULL;
GstRTSPTransport *video_transport = NULL;
GSocket *rtp_socket, *rtcp_socket;
GstRTSPClient *client;
GstRTSPClientClass *klass;
client = gst_rtsp_client_new ();
klass = GST_RTSP_CLIENT_GET_CLASS (client);
klass->adjust_play_mode = adjust_play_mode;
mounts = gst_rtsp_server_get_mount_points (server);
factory = gst_rtsp_media_factory_new ();
gst_rtsp_media_factory_set_launch (factory,
"( " VIDEO_PIPELINE " " AUDIO_PIPELINE " )");
gst_rtsp_mount_points_add_factory (mounts, TEST_MOUNT_POINT, factory);
g_object_unref (mounts);
/* set port to any */
gst_rtsp_server_set_service (server, "0");
/* attach to default main context */
source_id = gst_rtsp_server_attach (server, NULL);
fail_if (source_id == 0);
/* get port */
service = gst_rtsp_server_get_service (server);
test_port = atoi (service);
fail_unless (test_port != 0);
g_free (service);
conn = connect_to_server (test_port, TEST_MOUNT_POINT);
sdp_message = do_describe (conn, TEST_MOUNT_POINT);
/* get control strings from DESCRIBE response */
fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
sdp_media = gst_sdp_message_get_media (sdp_message, 0);
video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
sdp_media = gst_sdp_message_get_media (sdp_message, 1);
audio_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
get_client_ports_full (&client_port, &rtp_socket, &rtcp_socket);
/* do SETUP for video */
fail_unless (do_setup (conn, video_control, &client_port, &session,
&video_transport) == GST_RTSP_STS_OK);
/* do SETUP for audio */
fail_unless (do_setup (conn, audio_control, &client_port, &session,
&audio_transport) == GST_RTSP_STS_OK);
/* send PLAY request and check that we get 200 OK */
fail_unless (do_simple_request_rangein (conn, GST_RTSP_PLAY,
session, "npt=0-") == GST_RTSP_STS_OK);
/* let it play for a while, so it needs to seek
* for next play (npt=0-) */
g_usleep (30000);
/* send PLAY request and check that we get 200 OK */
fail_unless (do_simple_request_rangein (conn, GST_RTSP_PLAY,
session, "npt=0-") == GST_RTSP_STS_OK);
/* send TEARDOWN request and check that we get 200 OK */
fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
session) == GST_RTSP_STS_OK);
/* clean up and iterate so the clean-up can finish */
g_free (session);
gst_rtsp_transport_free (video_transport);
gst_rtsp_transport_free (audio_transport);
gst_sdp_message_free (sdp_message);
gst_rtsp_connection_free (conn);
stop_server ();
iterate ();
}
GST_END_TEST;
static Suite *
rtspserver_suite (void)
{
@ -2512,6 +2637,7 @@ rtspserver_suite (void)
tcase_add_test (tc, test_record_tcp);
tcase_add_test (tc, test_multiple_transports);
tcase_add_test (tc, test_suspend_mode_reset_only_audio);
tcase_add_test (tc, test_double_play);
return s;
}