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client: cleanups

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Rename variables for clarity
Keep media in state when we can
This commit is contained in:
Wim Taymans 2013-07-02 11:58:02 +02:00
parent a7fe63298c
commit 0248775c74
1 changed files with 35 additions and 36 deletions
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71
gst/rtsp-server/rtsp-client.c
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@ -105,7 +105,7 @@ static GstSDPMessage *create_sdp (GstRTSPClient * client, GstRTSPMedia * media);
static void client_session_finalized (GstRTSPClient * client, static void client_session_finalized (GstRTSPClient * client,
GstRTSPSession * session); GstRTSPSession * session);
static void unlink_session_transports (GstRTSPClient * client, static void unlink_session_transports (GstRTSPClient * client,
GstRTSPSession * session, GstRTSPSessionMedia * media); GstRTSPSession * session, GstRTSPSessionMedia * sessmedia);
static gboolean default_configure_client_transport (GstRTSPClient * client, static gboolean default_configure_client_transport (GstRTSPClient * client,
GstRTSPClientState * state, GstRTSPTransport * ct); GstRTSPClientState * state, GstRTSPTransport * ct);
static GstRTSPResult default_params_set (GstRTSPClient * client, static GstRTSPResult default_params_set (GstRTSPClient * client,
@ -230,13 +230,13 @@ gst_rtsp_client_init (GstRTSPClient * client)
} }
static GstRTSPFilterResult static GstRTSPFilterResult
filter_session (GstRTSPSession * sess, GstRTSPSessionMedia * media, filter_session (GstRTSPSession * sess, GstRTSPSessionMedia * sessmedia,
gpointer user_data) gpointer user_data)
{ {
GstRTSPClient *client = GST_RTSP_CLIENT (user_data); GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
gst_rtsp_session_media_set_state (media, GST_STATE_NULL); gst_rtsp_session_media_set_state (sessmedia, GST_STATE_NULL);
unlink_session_transports (client, sess, media); unlink_session_transports (client, sess, sessmedia);
/* unmanage the media in the session */ /* unmanage the media in the session */
return GST_RTSP_FILTER_REMOVE; return GST_RTSP_FILTER_REMOVE;
@ -649,18 +649,18 @@ unlink_transport (GstRTSPClient * client, GstRTSPSession * session,
static void static void
unlink_session_transports (GstRTSPClient * client, GstRTSPSession * session, unlink_session_transports (GstRTSPClient * client, GstRTSPSession * session,
GstRTSPSessionMedia * media) GstRTSPSessionMedia * sessmedia)
{ {
guint n_streams, i; guint n_streams, i;
n_streams = n_streams =
gst_rtsp_media_n_streams (gst_rtsp_session_media_get_media (media)); gst_rtsp_media_n_streams (gst_rtsp_session_media_get_media (sessmedia));
for (i = 0; i < n_streams; i++) { for (i = 0; i < n_streams; i++) {
GstRTSPStreamTransport *trans; GstRTSPStreamTransport *trans;
const GstRTSPTransport *tr; const GstRTSPTransport *tr;
/* get the transport, if there is no transport configured, skip this stream */ /* get the transport, if there is no transport configured, skip this stream */
trans = gst_rtsp_session_media_get_transport (media, i); trans = gst_rtsp_session_media_get_transport (sessmedia, i);
if (trans == NULL) if (trans == NULL)
continue; continue;
@ -696,7 +696,7 @@ handle_teardown_request (GstRTSPClient * client, GstRTSPClientState * state)
{ {
GstRTSPClientPrivate *priv = client->priv; GstRTSPClientPrivate *priv = client->priv;
GstRTSPSession *session; GstRTSPSession *session;
GstRTSPSessionMedia *media; GstRTSPSessionMedia *sessmedia;
GstRTSPStatusCode code; GstRTSPStatusCode code;
if (!state->session) if (!state->session)
@ -708,27 +708,27 @@ handle_teardown_request (GstRTSPClient * client, GstRTSPClientState * state)
goto no_uri; goto no_uri;
/* get a handle to the configuration of the media in the session */ /* get a handle to the configuration of the media in the session */
media = gst_rtsp_session_get_media (session, state->uri); sessmedia = gst_rtsp_session_get_media (session, state->uri);
if (!media) if (!sessmedia)
goto not_found; goto not_found;
state->sessmedia = media; state->sessmedia = sessmedia;
/* we emit the signal before closing the connection */ /* we emit the signal before closing the connection */
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST], g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST],
0, state); 0, state);
/* unlink the all TCP callbacks */ /* unlink the all TCP callbacks */
unlink_session_transports (client, session, media); unlink_session_transports (client, session, sessmedia);
/* remove the session from the watched sessions */ /* remove the session from the watched sessions */
client_unwatch_session (client, session); client_unwatch_session (client, session);
gst_rtsp_session_media_set_state (media, GST_STATE_NULL); gst_rtsp_session_media_set_state (sessmedia, GST_STATE_NULL);
/* unmanage the media in the session, returns false if all media session /* unmanage the media in the session, returns false if all media session
* are torn down. */ * are torn down. */
if (!gst_rtsp_session_release_media (session, media)) { if (!gst_rtsp_session_release_media (session, sessmedia)) {
/* remove the session */ /* remove the session */
gst_rtsp_session_pool_remove (priv->session_pool, session); gst_rtsp_session_pool_remove (priv->session_pool, session);
} }
@ -860,7 +860,7 @@ static gboolean
handle_pause_request (GstRTSPClient * client, GstRTSPClientState * state) handle_pause_request (GstRTSPClient * client, GstRTSPClientState * state)
{ {
GstRTSPSession *session; GstRTSPSession *session;
GstRTSPSessionMedia *media; GstRTSPSessionMedia *sessmedia;
GstRTSPStatusCode code; GstRTSPStatusCode code;
GstRTSPState rtspstate; GstRTSPState rtspstate;
@ -871,23 +871,23 @@ handle_pause_request (GstRTSPClient * client, GstRTSPClientState * state)
goto no_uri; goto no_uri;
/* get a handle to the configuration of the media in the session */ /* get a handle to the configuration of the media in the session */
media = gst_rtsp_session_get_media (session, state->uri); sessmedia = gst_rtsp_session_get_media (session, state->uri);
if (!media) if (!sessmedia)
goto not_found; goto not_found;
state->sessmedia = media; state->sessmedia = sessmedia;
rtspstate = gst_rtsp_session_media_get_rtsp_state (media); rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
/* the session state must be playing or recording */ /* the session state must be playing or recording */
if (rtspstate != GST_RTSP_STATE_PLAYING && if (rtspstate != GST_RTSP_STATE_PLAYING &&
rtspstate != GST_RTSP_STATE_RECORDING) rtspstate != GST_RTSP_STATE_RECORDING)
goto invalid_state; goto invalid_state;
/* unlink the all TCP callbacks */ /* unlink the all TCP callbacks */
unlink_session_transports (client, session, media); unlink_session_transports (client, session, sessmedia);
/* then pause sending */ /* then pause sending */
gst_rtsp_session_media_set_state (media, GST_STATE_PAUSED); gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PAUSED);
/* construct the response now */ /* construct the response now */
code = GST_RTSP_STS_OK; code = GST_RTSP_STS_OK;
@ -897,7 +897,7 @@ handle_pause_request (GstRTSPClient * client, GstRTSPClientState * state)
send_message (client, session, state->response, FALSE); send_message (client, session, state->response, FALSE);
/* the state is now READY */ /* the state is now READY */
gst_rtsp_session_media_set_rtsp_state (media, GST_RTSP_STATE_READY); gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_READY);
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST], g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST],
0, state); 0, state);
@ -936,7 +936,8 @@ static gboolean
handle_play_request (GstRTSPClient * client, GstRTSPClientState * state) handle_play_request (GstRTSPClient * client, GstRTSPClientState * state)
{ {
GstRTSPSession *session; GstRTSPSession *session;
GstRTSPSessionMedia *media; GstRTSPSessionMedia *sessmedia;
GstRTSPMedia *media;
GstRTSPStatusCode code; GstRTSPStatusCode code;
GString *rtpinfo; GString *rtpinfo;
guint n_streams, i, infocount; guint n_streams, i, infocount;
@ -953,14 +954,15 @@ handle_play_request (GstRTSPClient * client, GstRTSPClientState * state)
goto no_uri; goto no_uri;
/* get a handle to the configuration of the media in the session */ /* get a handle to the configuration of the media in the session */
media = gst_rtsp_session_get_media (session, state->uri); sessmedia = gst_rtsp_session_get_media (session, state->uri);
if (!media) if (!sessmedia)
goto not_found; goto not_found;
state->sessmedia = media; state->sessmedia = sessmedia;
state->media = media = gst_rtsp_session_media_get_media (sessmedia);
/* the session state must be playing or ready */ /* the session state must be playing or ready */
rtspstate = gst_rtsp_session_media_get_rtsp_state (media); rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
if (rtspstate != GST_RTSP_STATE_PLAYING && rtspstate != GST_RTSP_STATE_READY) if (rtspstate != GST_RTSP_STATE_PLAYING && rtspstate != GST_RTSP_STATE_READY)
goto invalid_state; goto invalid_state;
@ -971,7 +973,7 @@ handle_play_request (GstRTSPClient * client, GstRTSPClientState * state)
if (gst_rtsp_range_parse (str, &range) == GST_RTSP_OK) { if (gst_rtsp_range_parse (str, &range) == GST_RTSP_OK) {
/* we have a range, seek to the position */ /* we have a range, seek to the position */
unit = range->unit; unit = range->unit;
gst_rtsp_media_seek (gst_rtsp_session_media_get_media (media), range); gst_rtsp_media_seek (media, range);
gst_rtsp_range_free (range); gst_rtsp_range_free (range);
} }
} }
@ -979,8 +981,7 @@ handle_play_request (GstRTSPClient * client, GstRTSPClientState * state)
/* grab RTPInfo from the payloaders now */ /* grab RTPInfo from the payloaders now */
rtpinfo = g_string_new (""); rtpinfo = g_string_new ("");
n_streams = n_streams = gst_rtsp_media_n_streams (media);
gst_rtsp_media_n_streams (gst_rtsp_session_media_get_media (media));
for (i = 0, infocount = 0; i < n_streams; i++) { for (i = 0, infocount = 0; i < n_streams; i++) {
GstRTSPStreamTransport *trans; GstRTSPStreamTransport *trans;
GstRTSPStream *stream; GstRTSPStream *stream;
@ -989,7 +990,7 @@ handle_play_request (GstRTSPClient * client, GstRTSPClientState * state)
guint rtptime, seq; guint rtptime, seq;
/* get the transport, if there is no transport configured, skip this stream */ /* get the transport, if there is no transport configured, skip this stream */
trans = gst_rtsp_session_media_get_transport (media, i); trans = gst_rtsp_session_media_get_transport (sessmedia, i);
if (trans == NULL) { if (trans == NULL) {
GST_INFO ("stream %d is not configured", i); GST_INFO ("stream %d is not configured", i);
continue; continue;
@ -1031,17 +1032,15 @@ handle_play_request (GstRTSPClient * client, GstRTSPClientState * state)
} }
/* add the range */ /* add the range */
str = str = gst_rtsp_media_get_range_string (media, TRUE, unit);
gst_rtsp_media_get_range_string (gst_rtsp_session_media_get_media (media),
TRUE, unit);
gst_rtsp_message_take_header (state->response, GST_RTSP_HDR_RANGE, str); gst_rtsp_message_take_header (state->response, GST_RTSP_HDR_RANGE, str);
send_message (client, session, state->response, FALSE); send_message (client, session, state->response, FALSE);
/* start playing after sending the request */ /* start playing after sending the request */
gst_rtsp_session_media_set_state (media, GST_STATE_PLAYING); gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PLAYING);
gst_rtsp_session_media_set_rtsp_state (media, GST_RTSP_STATE_PLAYING); gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_PLAYING);
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST], g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST],
0, state); 0, state);