mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2025-02-17 11:45:25 +00:00
rtspsrc: Rework the async state handling
Always send the flushing events to the udp elements now that basesrc supports this. This makes sure a segment event is sent correctly after a flush. Keep track of the currently executing command and make it possible to specify what command you want to cancel when starting a new async command. See https://bugzilla.gnome.org/show_bug.cgi?id=677905
This commit is contained in:
Wim Taymans
parent
ea17c457f9
commit
935472aba7
2 changed files with 54 additions and 66 deletions
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@ -248,15 +248,14 @@ static void gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message);
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static gboolean gst_rtspsrc_setup_auth (GstRTSPSrc * src,
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GstRTSPMessage * response);
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static void gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd);
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static void gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd, gint mask);
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static GstRTSPResult gst_rtspsrc_send_cb (GstRTSPExtension * ext,
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GstRTSPMessage * request, GstRTSPMessage * response, GstRTSPSrc * src);
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static GstRTSPResult gst_rtspsrc_open (GstRTSPSrc * src, gboolean async);
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static GstRTSPResult gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment,
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gboolean async);
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static GstRTSPResult gst_rtspsrc_pause (GstRTSPSrc * src, gboolean idle,
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gboolean async);
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static GstRTSPResult gst_rtspsrc_pause (GstRTSPSrc * src, gboolean async);
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static GstRTSPResult gst_rtspsrc_close (GstRTSPSrc * src, gboolean async,
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gboolean only_close);
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@ -266,18 +265,17 @@ static gboolean gst_rtspsrc_uri_set_uri (GstURIHandler * handler,
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static gboolean gst_rtspsrc_activate_streams (GstRTSPSrc * src);
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static gboolean gst_rtspsrc_loop (GstRTSPSrc * src);
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static gboolean gst_rtspsrc_stream_push_event (GstRTSPSrc * src,
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GstRTSPStream * stream, GstEvent * event, gboolean source);
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static gboolean gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event,
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gboolean source);
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GstRTSPStream * stream, GstEvent * event);
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static gboolean gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event);
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/* commands we send to out loop to notify it of events */
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#define CMD_OPEN 0
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#define CMD_PLAY 1
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#define CMD_PAUSE 2
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#define CMD_CLOSE 3
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#define CMD_WAIT 4
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#define CMD_RECONNECT 5
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#define CMD_LOOP 6
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#define CMD_OPEN (1 << 0)
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#define CMD_PLAY (1 << 1)
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#define CMD_PAUSE (1 << 2)
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#define CMD_CLOSE (1 << 3)
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#define CMD_WAIT (1 << 4)
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#define CMD_RECONNECT (1 << 5)
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#define CMD_LOOP (1 << 6)
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#define GST_ELEMENT_PROGRESS(el, type, code, text) \
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G_STMT_START { \
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@ -1688,8 +1686,8 @@ gst_rtspsrc_flush (GstRTSPSrc * src, gboolean flush, gboolean playing)
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gst_object_unref (clock);
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}
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}
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gst_rtspsrc_push_event (src, event, FALSE);
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gst_rtspsrc_loop_send_cmd (src, cmd);
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gst_rtspsrc_push_event (src, event);
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gst_rtspsrc_loop_send_cmd (src, cmd, CMD_LOOP);
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/* set up manager before data-flow resumes */
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/* to manage jitterbuffer buffer mode */
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@ -1866,7 +1864,7 @@ gst_rtspsrc_perform_seek (GstRTSPSrc * src, GstEvent * event)
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if (playing) {
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/* obtain current position in case seek fails */
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gst_rtspsrc_get_position (src);
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gst_rtspsrc_pause (src, FALSE, FALSE);
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gst_rtspsrc_pause (src, FALSE);
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}
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src->skip = skip;
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@ -2291,7 +2289,7 @@ gst_rtspsrc_do_stream_eos (GstRTSPSrc * src, GstRTSPStream * stream)
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goto was_eos;
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stream->eos = TRUE;
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gst_rtspsrc_stream_push_event (src, stream, gst_event_new_eos (), TRUE);
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gst_rtspsrc_stream_push_event (src, stream, gst_event_new_eos ());
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return;
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/* ERRORS */
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@ -3215,7 +3213,7 @@ done:
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static gboolean
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gst_rtspsrc_stream_push_event (GstRTSPSrc * src, GstRTSPStream * stream,
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GstEvent * event, gboolean source)
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GstEvent * event)
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{
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gboolean res = TRUE;
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@ -3223,7 +3221,7 @@ gst_rtspsrc_stream_push_event (GstRTSPSrc * src, GstRTSPStream * stream,
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if (stream->srcpad == NULL)
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goto done;
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if (source && stream->udpsrc[0]) {
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if (stream->udpsrc[0]) {
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gst_event_ref (event);
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res = gst_element_send_event (stream->udpsrc[0], event);
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} else if (stream->channelpad[0]) {
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@ -3234,7 +3232,7 @@ gst_rtspsrc_stream_push_event (GstRTSPSrc * src, GstRTSPStream * stream,
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res = gst_pad_send_event (stream->channelpad[0], event);
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}
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if (source && stream->udpsrc[1]) {
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if (stream->udpsrc[1]) {
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gst_event_ref (event);
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res &= gst_element_send_event (stream->udpsrc[1], event);
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} else if (stream->channelpad[1]) {
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@ -3252,7 +3250,7 @@ done:
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}
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static gboolean
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gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event, gboolean source)
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gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event)
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{
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GList *streams;
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gboolean res = TRUE;
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@ -3261,7 +3259,7 @@ gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event, gboolean source)
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GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
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gst_event_ref (event);
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res &= gst_rtspsrc_stream_push_event (src, ostream, event, source);
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res &= gst_rtspsrc_stream_push_event (src, ostream, event);
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}
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gst_event_unref (event);
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@ -4067,7 +4065,7 @@ gst_rtspsrc_loop_end_cmd (GstRTSPSrc * src, gint cmd, GstRTSPResult ret)
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}
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static void
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gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd)
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gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd, gint mask)
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{
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gint old;
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@ -4077,19 +4075,21 @@ gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd)
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GST_DEBUG_OBJECT (src, "sending cmd %d", cmd);
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GST_OBJECT_LOCK (src);
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old = src->loop_cmd;
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old = src->pending_cmd;
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if (old != CMD_WAIT) {
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src->loop_cmd = CMD_WAIT;
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src->pending_cmd = CMD_WAIT;
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GST_OBJECT_UNLOCK (src);
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/* cancel previous request */
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gst_rtspsrc_loop_cancel_cmd (src, old);
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GST_OBJECT_LOCK (src);
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}
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src->loop_cmd = cmd;
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src->pending_cmd = cmd;
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/* interrupt if allowed */
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if (src->waiting) {
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GST_DEBUG_OBJECT (src, "start connection flush");
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if (src->busy_cmd & mask) {
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GST_DEBUG_OBJECT (src, "connection flush busy %d", src->busy_cmd);
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gst_rtspsrc_connection_flush (src, TRUE);
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} else {
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GST_DEBUG_OBJECT (src, "not interrupting busy cmd %d", src->busy_cmd);
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}
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if (src->task)
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gst_task_start (src->task);
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@ -4134,7 +4134,7 @@ pause:
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gst_message_new_segment_done (GST_OBJECT_CAST (src),
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src->segment.format, src->segment.position));
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} else {
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gst_rtspsrc_push_event (src, gst_event_new_eos (), FALSE);
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gst_rtspsrc_push_event (src, gst_event_new_eos ());
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}
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} else if (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_EOS) {
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/* for fatal errors we post an error message, post the error before the
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@ -4142,7 +4142,7 @@ pause:
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GST_ELEMENT_ERROR (src, STREAM, FAILED,
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("Internal data flow error."),
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("streaming task paused, reason %s (%d)", reason, ret));
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gst_rtspsrc_push_event (src, gst_event_new_eos (), FALSE);
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gst_rtspsrc_push_event (src, gst_event_new_eos ());
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}
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return FALSE;
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}
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@ -6252,7 +6252,7 @@ send_error:
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}
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static GstRTSPResult
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gst_rtspsrc_pause (GstRTSPSrc * src, gboolean idle, gboolean async)
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gst_rtspsrc_pause (GstRTSPSrc * src, gboolean async)
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{
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GstRTSPResult res = GST_RTSP_OK;
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GstRTSPMessage request = { 0 };
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@ -6399,7 +6399,7 @@ gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message)
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/* we only act on the first udp timeout message, others are irrelevant
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* and can be ignored. */
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if (!ignore_timeout)
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gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_RECONNECT);
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gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_RECONNECT, CMD_LOOP);
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/* eat and free */
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gst_message_unref (message);
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return;
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@ -6456,46 +6456,36 @@ static void
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gst_rtspsrc_thread (GstRTSPSrc * src)
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{
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gint cmd;
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GstRTSPResult ret;
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gboolean running = FALSE;
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GST_OBJECT_LOCK (src);
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cmd = src->loop_cmd;
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src->loop_cmd = CMD_WAIT;
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cmd = src->pending_cmd;
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src->pending_cmd = CMD_WAIT;
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GST_DEBUG_OBJECT (src, "got command %d", cmd);
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/* we got the message command, so ensure communication is possible again */
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gst_rtspsrc_connection_flush (src, FALSE);
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/* we allow these to be interrupted */
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if (cmd == CMD_LOOP || cmd == CMD_CLOSE || cmd == CMD_PAUSE)
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src->waiting = TRUE;
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src->busy_cmd = cmd;
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GST_OBJECT_UNLOCK (src);
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switch (cmd) {
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case CMD_OPEN:
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ret = gst_rtspsrc_open (src, TRUE);
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gst_rtspsrc_open (src, TRUE);
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break;
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case CMD_PLAY:
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ret = gst_rtspsrc_play (src, &src->segment, TRUE);
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if (ret == GST_RTSP_OK)
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running = TRUE;
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gst_rtspsrc_play (src, &src->segment, TRUE);
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break;
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case CMD_PAUSE:
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ret = gst_rtspsrc_pause (src, TRUE, TRUE);
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if (ret == GST_RTSP_OK)
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running = TRUE;
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gst_rtspsrc_pause (src, TRUE);
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break;
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case CMD_CLOSE:
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ret = gst_rtspsrc_close (src, TRUE, FALSE);
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gst_rtspsrc_close (src, TRUE, FALSE);
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break;
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case CMD_LOOP:
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running = gst_rtspsrc_loop (src);
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gst_rtspsrc_loop (src);
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break;
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case CMD_RECONNECT:
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ret = gst_rtspsrc_reconnect (src, FALSE);
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if (ret == GST_RTSP_OK)
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running = TRUE;
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gst_rtspsrc_reconnect (src, FALSE);
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break;
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default:
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break;
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@ -6503,14 +6493,12 @@ gst_rtspsrc_thread (GstRTSPSrc * src)
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GST_OBJECT_LOCK (src);
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/* and go back to sleep */
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if (src->loop_cmd == CMD_WAIT) {
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if (running)
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src->loop_cmd = CMD_LOOP;
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else if (src->task)
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if (src->pending_cmd == CMD_WAIT) {
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if (src->task)
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gst_task_pause (src->task);
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}
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/* reset waiting */
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src->waiting = FALSE;
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src->busy_cmd = CMD_WAIT;
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GST_OBJECT_UNLOCK (src);
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}
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@ -6521,7 +6509,7 @@ gst_rtspsrc_start (GstRTSPSrc * src)
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GST_OBJECT_LOCK (src);
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src->loop_cmd = CMD_WAIT;
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src->pending_cmd = CMD_WAIT;
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if (src->task == NULL) {
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src->task = gst_task_new ((GstTaskFunction) gst_rtspsrc_thread, src);
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@ -6550,7 +6538,7 @@ gst_rtspsrc_stop (GstRTSPSrc * src)
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GST_DEBUG_OBJECT (src, "stopping");
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/* also cancels pending task */
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gst_rtspsrc_loop_send_cmd (src, CMD_WAIT);
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gst_rtspsrc_loop_send_cmd (src, CMD_WAIT, CMD_CLOSE);
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GST_OBJECT_LOCK (src);
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if ((task = src->task)) {
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@ -6598,12 +6586,12 @@ gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
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/* first attempt, don't ignore timeouts */
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rtspsrc->ignore_timeout = FALSE;
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rtspsrc->open_error = FALSE;
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gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_OPEN);
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gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_OPEN, 0);
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break;
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case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
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case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
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/* unblock the tcp tasks and make the loop waiting */
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gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_WAIT);
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gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_WAIT, CMD_LOOP);
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break;
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case GST_STATE_CHANGE_PAUSED_TO_READY:
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break;
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@ -6617,18 +6605,18 @@ gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
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switch (transition) {
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case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
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gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PLAY);
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gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PLAY, 0);
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break;
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case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
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/* send pause request and keep the idle task around */
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gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PAUSE);
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gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PAUSE, CMD_LOOP);
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ret = GST_STATE_CHANGE_NO_PREROLL;
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break;
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case GST_STATE_CHANGE_READY_TO_PAUSED:
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ret = GST_STATE_CHANGE_NO_PREROLL;
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break;
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case GST_STATE_CHANGE_PAUSED_TO_READY:
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gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_CLOSE);
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gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_CLOSE, CMD_PAUSE);
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break;
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case GST_STATE_CHANGE_READY_TO_NULL:
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gst_rtspsrc_stop (rtspsrc);
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@ -6656,7 +6644,7 @@ gst_rtspsrc_send_event (GstElement * element, GstEvent * event)
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rtspsrc = GST_RTSPSRC (element);
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if (GST_EVENT_IS_DOWNSTREAM (event)) {
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res = gst_rtspsrc_push_event (rtspsrc, event, TRUE);
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res = gst_rtspsrc_push_event (rtspsrc, event);
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} else {
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res = GST_ELEMENT_CLASS (parent_class)->send_event (element, event);
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}
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@ -178,9 +178,9 @@ struct _GstRTSPSrc {
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GstClockTime base_time;
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/* UDP mode loop */
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gint loop_cmd;
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gint pending_cmd;
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gint busy_cmd;
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gboolean ignore_timeout;
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gboolean waiting;
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gboolean open_error;
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/* mutex for protecting state changes */
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