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rtspclientsink: Use the new rtsp-stream API

Browse source 
https://bugzilla.gnome.org/show_bug.cgi?id=790412
This commit is contained in:
Patricia Muscalu 2017-12-16 21:46:53 +01:00 committed by Edward Hervey
parent 96cfed48bf
commit 64f1a3ab85
2 changed files with 75 additions and 13 deletions
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83
gst/rtsp-sink/gstrtspclientsink.c
View file

@ -3037,10 +3037,6 @@ gst_rtsp_client_sink_open (GstRTSPClientSink * sink, gboolean async)
if (async)
gst_rtsp_client_sink_loop_end_cmd (sink, CMD_OPEN, ret);
/* Collect all our input streams and create
* stream objects before actually returning */
gst_rtsp_client_sink_collect_streams (sink);
return ret;
/* ERRORS */
@ -3357,6 +3353,9 @@ gst_rtsp_client_sink_collect_streams (GstRTSPClientSink * sink)
}
context->joined = TRUE;
/* Block the stream, as it does not have any transport parts yet */
gst_rtsp_stream_set_blocked (context->stream, TRUE);
/* Let the stream object receive data */
gst_pad_remove_probe (srcpad, context->payloader_block_id);
@ -3710,6 +3709,28 @@ gst_rtsp_client_sink_setup_streams (GstRTSPClientSink * sink, gboolean async)
GST_ELEMENT_PROGRESS (sink, CONTINUE, "request", ("SETUP stream %d",
context->index));
{
GstRTSPTransport *transport;
gst_rtsp_transport_new (&transport);
if (gst_rtsp_transport_parse (transports, transport) != GST_RTSP_OK)
goto parse_transport_failed;
if (transport->lower_transport != GST_RTSP_LOWER_TRANS_TCP) {
if (!gst_rtsp_stream_allocate_udp_sockets (stream, family, transport,
FALSE)) {
gst_rtsp_transport_free (transport);
goto allocate_udp_ports_failed;
}
}
if (!gst_rtsp_stream_complete_stream (stream, transport)) {
gst_rtsp_transport_free (transport);
goto complete_stream_failed;
}
gst_rtsp_transport_free (transport);
gst_rtsp_stream_set_blocked (stream, FALSE);
}
/* handle the code ourselves */
res = gst_rtsp_client_sink_send (sink, info, &request, &response, &code);
if (res < 0)
@ -3885,6 +3906,30 @@ create_request_failed:
g_free (str);
goto cleanup_error;
}
parse_transport_failed:
{
GST_RTSP_STATE_UNLOCK (sink);
GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
("Could not parse transport."));
res = GST_RTSP_ERROR;
goto cleanup_error;
}
allocate_udp_ports_failed:
{
GST_RTSP_STATE_UNLOCK (sink);
GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
("Could not parse transport."));
res = GST_RTSP_ERROR;
goto cleanup_error;
}
complete_stream_failed:
{
GST_RTSP_STATE_UNLOCK (sink);
GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
("Could not parse transport."));
res = GST_RTSP_ERROR;
goto cleanup_error;
}
send_error:
{
gchar *str = gst_rtsp_strresult (res);
@ -3958,19 +4003,28 @@ gst_rtsp_client_sink_record (GstRTSPClientSink * sink, gboolean async)
GSocket *conn_socket;
GList *walk;
/* Wait for streams to preroll */
g_mutex_lock (&sink->preroll_lock);
while (sink->in_async) {
GST_LOG_OBJECT (sink, "Waiting for ASYNC_DONE preroll");
g_cond_wait (&sink->preroll_cond, &sink->preroll_lock);
}
g_mutex_unlock (&sink->preroll_lock);
if (sink->state == GST_RTSP_STATE_PLAYING) {
/* Already recording, don't send another request */
GST_LOG_OBJECT (sink, "Already in RECORD. Skipping duplicate request.");
g_mutex_unlock (&sink->preroll_lock);
goto done;
}
g_mutex_unlock (&sink->preroll_lock);
/* Collect all our input streams and create
* stream objects before actually returning.
* The streams are blocked at this point as we do not have any transport
* parts yet. */
gst_rtsp_client_sink_collect_streams (sink);
g_mutex_lock (&sink->block_streams_lock);
/* Wait for streams to be blocked */
while (!sink->streams_blocked) {
GST_DEBUG_OBJECT (sink, "waiting for streams to be blocked");
g_cond_wait (&sink->block_streams_cond, &sink->block_streams_lock);
}
g_mutex_unlock (&sink->block_streams_lock);
/* Send announce, then setup for all streams */
gst_sdp_message_init (&sink->cursdp);
@ -4294,7 +4348,11 @@ gst_rtsp_client_sink_handle_message (GstBin * bin, GstMessage * message)
return;
} else if (gst_structure_has_name (s, "GstRTSPStreamBlocking")) {
/* An RTSPStream has prerolled */
g_cond_broadcast (&rtsp_client_sink->preroll_cond);
GST_DEBUG_OBJECT (rtsp_client_sink, "received GstRTSPStreamBlocking");
g_mutex_lock (&rtsp_client_sink->block_streams_lock);
rtsp_client_sink->streams_blocked = TRUE;
g_cond_broadcast (&rtsp_client_sink->block_streams_cond);
g_mutex_unlock (&rtsp_client_sink->block_streams_lock);
}
GST_BIN_CLASS (parent_class)->handle_message (bin, message);
break;
@ -4443,7 +4501,6 @@ gst_rtsp_client_sink_start (GstRTSPClientSink * sink)
GST_DEBUG_OBJECT (sink, "starting");
sink->streams_collected = FALSE;
sink->in_async = TRUE;
gst_element_set_locked_state (GST_ELEMENT (sink->internal_bin), TRUE);
gst_rtsp_client_sink_set_state (sink, GST_STATE_READY);

5
gst/rtsp-sink/gstrtspclientsink.h
View file

@ -217,6 +217,11 @@ struct _GstRTSPClientSink {
/* TRUE when stream info has been collected */
gboolean streams_collected;
/* TRUE when streams have been blocked */
gboolean streams_blocked;
GMutex block_streams_lock;
GCond block_streams_cond;
guint next_pad_id;
gint next_dyn_pt;