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synced 2024-10-06 10:42:22 +00:00
rtsp-media: Wait on async when needed.
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.
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7e1edcf1a4
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18f4f4e509
2 changed files with 169 additions and 4 deletions
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@ -152,6 +152,7 @@ struct _GstRTSPMediaPrivate
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/* Dynamic element handling */
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guint nb_dynamic_elements;
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guint no_more_pads_pending;
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gboolean expected_async_done;
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};
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#define DEFAULT_SHARED FALSE
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@ -476,6 +477,7 @@ gst_rtsp_media_init (GstRTSPMedia * media)
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priv->max_mcast_ttl = DEFAULT_MAX_MCAST_TTL;
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priv->bind_mcast_address = DEFAULT_BIND_MCAST_ADDRESS;
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priv->do_rate_control = DEFAULT_DO_RATE_CONTROL;
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priv->expected_async_done = FALSE;
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}
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static void
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@ -2819,6 +2821,30 @@ gst_rtsp_media_seek_trickmode (GstRTSPMedia * media,
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GstEvent *seek_event;
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gboolean unblock = FALSE;
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/* Handle expected async-done before waiting on next async-done.
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*
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* Since the seek further down in code will cause a preroll and
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* a async-done will be generated it's important to wait on async-done
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* if that is expected. Otherwise there is the risk that the waiting
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* for async-done after the seek is detecting the expected async-done
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* instead of the one that corresponds to the seek. Then execution
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* continue and act as if the pipeline is prerolled, but it's not.
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*
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* During wait_preroll message GST_MESSAGE_ASYNC_DONE will come
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* and then the state will change from preparing to prepared */
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if (priv->expected_async_done) {
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GST_DEBUG (" expected to get async-done, waiting ");
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gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
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g_rec_mutex_unlock (&priv->state_lock);
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/* wait until pipeline is prerolled */
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if (!wait_preroll (media))
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goto preroll_failed_expected_async_done;
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g_rec_mutex_lock (&priv->state_lock);
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GST_DEBUG (" got expected async-done");
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}
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gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
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if (rate < 0.0) {
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@ -2910,6 +2936,11 @@ preroll_failed:
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GST_WARNING ("failed to preroll after seek");
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return FALSE;
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}
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preroll_failed_expected_async_done:
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{
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GST_WARNING ("failed to preroll");
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return FALSE;
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}
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}
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/**
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@ -3118,6 +3149,8 @@ default_handle_message (GstRTSPMedia * media, GstMessage * message)
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case GST_MESSAGE_STREAM_STATUS:
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break;
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case GST_MESSAGE_ASYNC_DONE:
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if (priv->expected_async_done)
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priv->expected_async_done = FALSE;
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if (priv->complete) {
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/* receive the final ASYNC_DONE, that is posted by the media pipeline
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* after all the transport parts have been successfully added to
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@ -4459,6 +4492,8 @@ static void
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media_set_pipeline_state_locked (GstRTSPMedia * media, GstState state)
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{
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GstRTSPMediaPrivate *priv = media->priv;
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GstStateChangeReturn set_state_ret;
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priv->expected_async_done = FALSE;
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if (state == GST_STATE_NULL) {
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gst_rtsp_media_unprepare (media);
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@ -4474,11 +4509,15 @@ media_set_pipeline_state_locked (GstRTSPMedia * media, GstState state)
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/* make sure pads are not blocking anymore when going to PLAYING */
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media_unblock_linked (media);
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set_state (media, state);
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if (state == GST_STATE_PAUSED) {
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set_state_ret = set_state (media, state);
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if (set_state_ret == GST_STATE_CHANGE_ASYNC)
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priv->expected_async_done = TRUE;
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/* and suspend after pause */
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if (state == GST_STATE_PAUSED)
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gst_rtsp_media_suspend (media);
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} else {
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set_state (media, state);
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}
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}
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}
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}
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@ -483,6 +483,16 @@ do_simple_request (GstRTSPConnection * conn, GstRTSPMethod method,
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NULL, NULL, NULL, NULL, NULL);
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}
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/* send an rtsp request with a method,session and range in,
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* and receive response. range_in is the Range in req header */
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static GstRTSPStatusCode
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do_simple_request_rangein (GstRTSPConnection * conn, GstRTSPMethod method,
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const gchar * session, const gchar * rangein)
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{
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return do_request (conn, method, NULL, session, NULL, rangein, NULL,
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NULL, NULL, NULL, NULL, NULL);
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}
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/* send a DESCRIBE request and receive response. returns a received
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* GstSDPMessage that must be freed by the caller */
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static GstSDPMessage *
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@ -2473,6 +2483,121 @@ GST_START_TEST (test_suspend_mode_reset_only_audio)
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GST_END_TEST;
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static GstRTSPStatusCode
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adjust_play_mode (GstRTSPClient * client, GstRTSPContext * ctx,
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GstRTSPTimeRange ** range, GstSeekFlags * flags, gdouble * rate,
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GstClockTime * trickmode_interval, gboolean * enable_rate_control)
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{
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GstRTSPState rtspstate;
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rtspstate = gst_rtsp_session_media_get_rtsp_state (ctx->sessmedia);
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if (rtspstate == GST_RTSP_STATE_PLAYING) {
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if (!gst_rtsp_session_media_set_state (ctx->sessmedia, GST_STATE_PAUSED))
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return GST_RTSP_STS_INTERNAL_SERVER_ERROR;
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if (!gst_rtsp_media_unsuspend (ctx->media))
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return GST_RTSP_STS_INTERNAL_SERVER_ERROR;
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}
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return GST_RTSP_STS_OK;
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}
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GST_START_TEST (test_double_play)
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{
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GstRTSPMountPoints *mounts;
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gchar *service;
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GstRTSPMediaFactory *factory;
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GstRTSPConnection *conn;
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GstSDPMessage *sdp_message = NULL;
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const GstSDPMedia *sdp_media;
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const gchar *video_control;
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const gchar *audio_control;
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GstRTSPRange client_port;
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gchar *session = NULL;
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GstRTSPTransport *audio_transport = NULL;
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GstRTSPTransport *video_transport = NULL;
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GSocket *rtp_socket, *rtcp_socket;
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GstRTSPClient *client;
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GstRTSPClientClass *klass;
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client = gst_rtsp_client_new ();
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klass = GST_RTSP_CLIENT_GET_CLASS (client);
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klass->adjust_play_mode = adjust_play_mode;
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mounts = gst_rtsp_server_get_mount_points (server);
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factory = gst_rtsp_media_factory_new ();
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gst_rtsp_media_factory_set_launch (factory,
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"( " VIDEO_PIPELINE " " AUDIO_PIPELINE " )");
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gst_rtsp_mount_points_add_factory (mounts, TEST_MOUNT_POINT, factory);
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g_object_unref (mounts);
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/* set port to any */
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gst_rtsp_server_set_service (server, "0");
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/* attach to default main context */
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source_id = gst_rtsp_server_attach (server, NULL);
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fail_if (source_id == 0);
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/* get port */
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service = gst_rtsp_server_get_service (server);
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test_port = atoi (service);
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fail_unless (test_port != 0);
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g_free (service);
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conn = connect_to_server (test_port, TEST_MOUNT_POINT);
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sdp_message = do_describe (conn, TEST_MOUNT_POINT);
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/* get control strings from DESCRIBE response */
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fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
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sdp_media = gst_sdp_message_get_media (sdp_message, 0);
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video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
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sdp_media = gst_sdp_message_get_media (sdp_message, 1);
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audio_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
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get_client_ports_full (&client_port, &rtp_socket, &rtcp_socket);
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/* do SETUP for video */
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fail_unless (do_setup (conn, video_control, &client_port, &session,
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&video_transport) == GST_RTSP_STS_OK);
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/* do SETUP for audio */
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fail_unless (do_setup (conn, audio_control, &client_port, &session,
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&audio_transport) == GST_RTSP_STS_OK);
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/* send PLAY request and check that we get 200 OK */
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fail_unless (do_simple_request_rangein (conn, GST_RTSP_PLAY,
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session, "npt=0-") == GST_RTSP_STS_OK);
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/* let it play for a while, so it needs to seek
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* for next play (npt=0-) */
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g_usleep (30000);
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/* send PLAY request and check that we get 200 OK */
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fail_unless (do_simple_request_rangein (conn, GST_RTSP_PLAY,
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session, "npt=0-") == GST_RTSP_STS_OK);
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/* send TEARDOWN request and check that we get 200 OK */
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fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
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session) == GST_RTSP_STS_OK);
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/* clean up and iterate so the clean-up can finish */
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g_free (session);
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gst_rtsp_transport_free (video_transport);
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gst_rtsp_transport_free (audio_transport);
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gst_sdp_message_free (sdp_message);
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gst_rtsp_connection_free (conn);
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stop_server ();
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iterate ();
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}
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GST_END_TEST;
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static Suite *
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rtspserver_suite (void)
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{
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@ -2512,6 +2637,7 @@ rtspserver_suite (void)
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tcase_add_test (tc, test_record_tcp);
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tcase_add_test (tc, test_multiple_transports);
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tcase_add_test (tc, test_suspend_mode_reset_only_audio);
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tcase_add_test (tc, test_double_play);
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return s;
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}
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