mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2025-03-28 11:55:39 +00:00
client: cleanups
Rename variables for clarity Keep media in state when we can
This commit is contained in:
Wim Taymans
parent
a7fe63298c
commit
0248775c74
1 changed files with 35 additions and 36 deletions
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@ -105,7 +105,7 @@ static GstSDPMessage *create_sdp (GstRTSPClient * client, GstRTSPMedia * media);
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static void client_session_finalized (GstRTSPClient * client,
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GstRTSPSession * session);
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static void unlink_session_transports (GstRTSPClient * client,
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GstRTSPSession * session, GstRTSPSessionMedia * media);
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GstRTSPSession * session, GstRTSPSessionMedia * sessmedia);
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static gboolean default_configure_client_transport (GstRTSPClient * client,
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GstRTSPClientState * state, GstRTSPTransport * ct);
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static GstRTSPResult default_params_set (GstRTSPClient * client,
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@ -230,13 +230,13 @@ gst_rtsp_client_init (GstRTSPClient * client)
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}
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static GstRTSPFilterResult
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filter_session (GstRTSPSession * sess, GstRTSPSessionMedia * media,
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filter_session (GstRTSPSession * sess, GstRTSPSessionMedia * sessmedia,
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gpointer user_data)
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{
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GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
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gst_rtsp_session_media_set_state (media, GST_STATE_NULL);
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unlink_session_transports (client, sess, media);
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gst_rtsp_session_media_set_state (sessmedia, GST_STATE_NULL);
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unlink_session_transports (client, sess, sessmedia);
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/* unmanage the media in the session */
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return GST_RTSP_FILTER_REMOVE;
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@ -649,18 +649,18 @@ unlink_transport (GstRTSPClient * client, GstRTSPSession * session,
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static void
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unlink_session_transports (GstRTSPClient * client, GstRTSPSession * session,
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GstRTSPSessionMedia * media)
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GstRTSPSessionMedia * sessmedia)
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{
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guint n_streams, i;
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n_streams =
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gst_rtsp_media_n_streams (gst_rtsp_session_media_get_media (media));
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gst_rtsp_media_n_streams (gst_rtsp_session_media_get_media (sessmedia));
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for (i = 0; i < n_streams; i++) {
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GstRTSPStreamTransport *trans;
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const GstRTSPTransport *tr;
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/* get the transport, if there is no transport configured, skip this stream */
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trans = gst_rtsp_session_media_get_transport (media, i);
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trans = gst_rtsp_session_media_get_transport (sessmedia, i);
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if (trans == NULL)
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continue;
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@ -696,7 +696,7 @@ handle_teardown_request (GstRTSPClient * client, GstRTSPClientState * state)
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{
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GstRTSPClientPrivate *priv = client->priv;
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GstRTSPSession *session;
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GstRTSPSessionMedia *media;
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GstRTSPSessionMedia *sessmedia;
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GstRTSPStatusCode code;
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if (!state->session)
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@ -708,27 +708,27 @@ handle_teardown_request (GstRTSPClient * client, GstRTSPClientState * state)
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goto no_uri;
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/* get a handle to the configuration of the media in the session */
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media = gst_rtsp_session_get_media (session, state->uri);
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if (!media)
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sessmedia = gst_rtsp_session_get_media (session, state->uri);
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if (!sessmedia)
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goto not_found;
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state->sessmedia = media;
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state->sessmedia = sessmedia;
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/* we emit the signal before closing the connection */
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g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST],
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0, state);
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/* unlink the all TCP callbacks */
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unlink_session_transports (client, session, media);
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unlink_session_transports (client, session, sessmedia);
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/* remove the session from the watched sessions */
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client_unwatch_session (client, session);
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gst_rtsp_session_media_set_state (media, GST_STATE_NULL);
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gst_rtsp_session_media_set_state (sessmedia, GST_STATE_NULL);
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/* unmanage the media in the session, returns false if all media session
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* are torn down. */
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if (!gst_rtsp_session_release_media (session, media)) {
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if (!gst_rtsp_session_release_media (session, sessmedia)) {
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/* remove the session */
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gst_rtsp_session_pool_remove (priv->session_pool, session);
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}
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@ -860,7 +860,7 @@ static gboolean
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handle_pause_request (GstRTSPClient * client, GstRTSPClientState * state)
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{
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GstRTSPSession *session;
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GstRTSPSessionMedia *media;
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GstRTSPSessionMedia *sessmedia;
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GstRTSPStatusCode code;
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GstRTSPState rtspstate;
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@ -871,23 +871,23 @@ handle_pause_request (GstRTSPClient * client, GstRTSPClientState * state)
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goto no_uri;
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/* get a handle to the configuration of the media in the session */
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media = gst_rtsp_session_get_media (session, state->uri);
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if (!media)
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sessmedia = gst_rtsp_session_get_media (session, state->uri);
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if (!sessmedia)
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goto not_found;
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state->sessmedia = media;
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state->sessmedia = sessmedia;
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rtspstate = gst_rtsp_session_media_get_rtsp_state (media);
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rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
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/* the session state must be playing or recording */
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if (rtspstate != GST_RTSP_STATE_PLAYING &&
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rtspstate != GST_RTSP_STATE_RECORDING)
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goto invalid_state;
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/* unlink the all TCP callbacks */
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unlink_session_transports (client, session, media);
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unlink_session_transports (client, session, sessmedia);
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/* then pause sending */
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gst_rtsp_session_media_set_state (media, GST_STATE_PAUSED);
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gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PAUSED);
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/* construct the response now */
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code = GST_RTSP_STS_OK;
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@ -897,7 +897,7 @@ handle_pause_request (GstRTSPClient * client, GstRTSPClientState * state)
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send_message (client, session, state->response, FALSE);
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/* the state is now READY */
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gst_rtsp_session_media_set_rtsp_state (media, GST_RTSP_STATE_READY);
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gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_READY);
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g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST],
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0, state);
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@ -936,7 +936,8 @@ static gboolean
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handle_play_request (GstRTSPClient * client, GstRTSPClientState * state)
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{
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GstRTSPSession *session;
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GstRTSPSessionMedia *media;
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GstRTSPSessionMedia *sessmedia;
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GstRTSPMedia *media;
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GstRTSPStatusCode code;
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GString *rtpinfo;
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guint n_streams, i, infocount;
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@ -953,14 +954,15 @@ handle_play_request (GstRTSPClient * client, GstRTSPClientState * state)
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goto no_uri;
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/* get a handle to the configuration of the media in the session */
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media = gst_rtsp_session_get_media (session, state->uri);
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if (!media)
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sessmedia = gst_rtsp_session_get_media (session, state->uri);
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if (!sessmedia)
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goto not_found;
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state->sessmedia = media;
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state->sessmedia = sessmedia;
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state->media = media = gst_rtsp_session_media_get_media (sessmedia);
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/* the session state must be playing or ready */
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rtspstate = gst_rtsp_session_media_get_rtsp_state (media);
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rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
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if (rtspstate != GST_RTSP_STATE_PLAYING && rtspstate != GST_RTSP_STATE_READY)
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goto invalid_state;
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@ -971,7 +973,7 @@ handle_play_request (GstRTSPClient * client, GstRTSPClientState * state)
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if (gst_rtsp_range_parse (str, &range) == GST_RTSP_OK) {
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/* we have a range, seek to the position */
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unit = range->unit;
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gst_rtsp_media_seek (gst_rtsp_session_media_get_media (media), range);
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gst_rtsp_media_seek (media, range);
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gst_rtsp_range_free (range);
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}
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}
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@ -979,8 +981,7 @@ handle_play_request (GstRTSPClient * client, GstRTSPClientState * state)
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/* grab RTPInfo from the payloaders now */
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rtpinfo = g_string_new ("");
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n_streams =
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gst_rtsp_media_n_streams (gst_rtsp_session_media_get_media (media));
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n_streams = gst_rtsp_media_n_streams (media);
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for (i = 0, infocount = 0; i < n_streams; i++) {
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GstRTSPStreamTransport *trans;
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GstRTSPStream *stream;
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@ -989,7 +990,7 @@ handle_play_request (GstRTSPClient * client, GstRTSPClientState * state)
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guint rtptime, seq;
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/* get the transport, if there is no transport configured, skip this stream */
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trans = gst_rtsp_session_media_get_transport (media, i);
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trans = gst_rtsp_session_media_get_transport (sessmedia, i);
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if (trans == NULL) {
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GST_INFO ("stream %d is not configured", i);
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continue;
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@ -1031,17 +1032,15 @@ handle_play_request (GstRTSPClient * client, GstRTSPClientState * state)
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}
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/* add the range */
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str =
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gst_rtsp_media_get_range_string (gst_rtsp_session_media_get_media (media),
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TRUE, unit);
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str = gst_rtsp_media_get_range_string (media, TRUE, unit);
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gst_rtsp_message_take_header (state->response, GST_RTSP_HDR_RANGE, str);
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send_message (client, session, state->response, FALSE);
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/* start playing after sending the request */
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gst_rtsp_session_media_set_state (media, GST_STATE_PLAYING);
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gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PLAYING);
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gst_rtsp_session_media_set_rtsp_state (media, GST_RTSP_STATE_PLAYING);
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gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_PLAYING);
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g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST],
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0, state);
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