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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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gst/rtpmanager/async_jitter_queue.c: Fix the case where the buffer underruns and does not block.
Original commit message from CVS: * gst/rtpmanager/async_jitter_queue.c: (signal_waiting_threads), (async_jitter_queue_pop_intern_unlocked): Fix the case where the buffer underruns and does not block. * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_base_init), (create_recv_rtcp), (create_send_rtp), (create_rtcp), (gst_rtp_bin_request_new_pad): Rename RTCP send pad, like in the session manager. Allow getting an RTCP pad for receiving even if we don't receive RTP. fix handling of send_rtp_src pad. * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain): When no pt map could be found, fall back to the sinkpad caps. * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_process_rtp), (gst_rtp_session_send_rtp), (create_recv_rtp_sink), (create_recv_rtcp_sink), (create_send_rtp_sink), (create_send_rtcp_src): Fix pad names. * gst/rtpmanager/rtpsession.c: (source_push_rtp), (rtp_session_create_source), (rtp_session_process_sr), (rtp_session_send_rtp), (session_start_rtcp): * gst/rtpmanager/rtpsession.h: Unlock session when performing a callback. Add callbacks for the internal session object. Fix sending of RTP packets. first attempt at adding NTP times in the SR packets. Small debug and doc improvements. * gst/rtpmanager/rtpsource.c: (rtp_source_send_rtp): Update stats for SR reports.
This commit is contained in:
Wim Taymans
Tim-Philipp Müller
parent
e6537bcd7c
commit
600afaaff9
7 changed files with 116 additions and 43 deletions
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@ -100,6 +100,7 @@ signal_waiting_threads (AsyncJitterQueue * queue)
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{
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if (async_jitter_queue_length_ts_units_unlocked (queue) >=
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queue->high_threshold * queue->max_queue_length) {
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GST_DEBUG ("stop buffering");
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queue->buffering = FALSE;
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}
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@ -473,6 +474,7 @@ async_jitter_queue_pop_intern_unlocked (AsyncJitterQueue * queue)
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{
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gpointer retval;
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GstBuffer *tail_buffer = NULL;
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guint tsunits;
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if (queue->pop_flushing)
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return NULL;
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@ -485,20 +487,27 @@ async_jitter_queue_pop_intern_unlocked (AsyncJitterQueue * queue)
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return NULL;
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}
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if (async_jitter_queue_length_ts_units_unlocked (queue) <=
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queue->low_threshold * queue->max_queue_length
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tsunits = async_jitter_queue_length_ts_units_unlocked (queue);
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GST_DEBUG ("tsunits %u, pops: %u, limit %d", tsunits, queue->pops_remaining,
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queue->low_threshold * queue->max_queue_length);
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if (tsunits <= queue->low_threshold * queue->max_queue_length
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&& queue->pops_remaining == 0) {
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if (!queue->buffering) {
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GST_DEBUG ("start buffering");
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queue->buffering = TRUE;
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queue->pops_remaining = queue->queue->length;
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} else {
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while (!g_queue_peek_tail (queue->queue) || queue->pop_blocking) {
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queue->waiting_threads++;
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g_cond_wait (queue->cond, queue->mutex);
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queue->waiting_threads--;
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if (queue->pop_flushing)
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return NULL;
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}
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}
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GST_DEBUG ("wait for data");
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while (!g_queue_peek_tail (queue->queue) || queue->pop_blocking) {
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queue->waiting_threads++;
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g_cond_wait (queue->cond, queue->mutex);
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queue->waiting_threads--;
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if (queue->pop_flushing)
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return NULL;
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}
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}
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@ -84,8 +84,8 @@ GST_STATIC_PAD_TEMPLATE ("recv_rtp_src_%d_%d_%d",
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GST_STATIC_CAPS ("application/x-rtp")
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);
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static GstStaticPadTemplate rtpbin_rtcp_src_template =
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GST_STATIC_PAD_TEMPLATE ("rtcp_src_%d",
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static GstStaticPadTemplate rtpbin_send_rtcp_src_template =
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GST_STATIC_PAD_TEMPLATE ("send_rtcp_src_%d",
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GST_PAD_SRC,
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GST_PAD_REQUEST,
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GST_STATIC_CAPS ("application/x-rtcp")
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@ -195,7 +195,7 @@ struct _GstRTPBinSession
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GstPad *recv_rtcp_src;
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GstPad *send_rtp_sink;
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GstPad *send_rtp_src;
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GstPad *rtcp_src;
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GstPad *send_rtcp_src;
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};
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/* find a session with the given id. Must be called with RTP_BIN_LOCK */
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@ -432,7 +432,7 @@ gst_rtp_bin_base_init (gpointer klass)
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&rtpbin_recv_rtp_src_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&rtpbin_rtcp_src_template));
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gst_static_pad_template_get (&rtpbin_send_rtcp_src_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&rtpbin_send_rtp_src_template));
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@ -795,10 +795,15 @@ create_recv_rtcp (GstRTPBin * rtpbin, GstPadTemplate * templ,
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GST_DEBUG_OBJECT (rtpbin, "finding session %d", sessid);
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/* get the session, it must exist or we error */
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/* get or create the session */
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session = find_session_by_id (rtpbin, sessid);
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if (!session)
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goto no_session;
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if (!session) {
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GST_DEBUG_OBJECT (rtpbin, "creating session %d", sessid);
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/* create session now */
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session = create_session (rtpbin, sessid);
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if (session == NULL)
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goto create_error;
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}
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/* check if pad was requested */
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if (session->recv_rtcp_sink != NULL)
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@ -841,9 +846,9 @@ no_name:
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g_warning ("rtpbin: invalid name given");
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return NULL;
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}
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no_session:
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create_error:
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{
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g_warning ("rtpbin: no session with id %d", sessid);
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/* create_session already warned */
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return NULL;
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}
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existed:
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@ -872,7 +877,7 @@ link_failed:
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static GstPad *
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create_send_rtp (GstRTPBin * rtpbin, GstPadTemplate * templ, const gchar * name)
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{
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GstPad *result, *srcpad, *srcghost;
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GstPad *result, *srcghost;
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gchar *gname;
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guint sessid;
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GstRTPBinSession *session;
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@ -895,7 +900,7 @@ create_send_rtp (GstRTPBin * rtpbin, GstPadTemplate * templ, const gchar * name)
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if (session->send_rtp_sink != NULL)
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goto existed;
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/* get recv_rtp pad and store */
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/* get send_rtp pad and store */
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session->send_rtp_sink =
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gst_element_get_request_pad (session->session, "send_rtp_sink");
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if (session->send_rtp_sink == NULL)
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@ -907,8 +912,9 @@ create_send_rtp (GstRTPBin * rtpbin, GstPadTemplate * templ, const gchar * name)
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gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), result);
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/* get srcpad */
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srcpad = gst_element_get_pad (session->session, "send_rtp_src");
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if (srcpad == NULL)
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session->send_rtp_src =
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gst_element_get_static_pad (session->session, "send_rtp_src");
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if (session->send_rtp_src == NULL)
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goto no_srcpad;
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/* ghost the new source pad */
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@ -916,7 +922,7 @@ create_send_rtp (GstRTPBin * rtpbin, GstPadTemplate * templ, const gchar * name)
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gname = g_strdup_printf ("send_rtp_src_%d", sessid);
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templ = gst_element_class_get_pad_template (klass, "send_rtp_src_%d");
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srcghost =
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gst_ghost_pad_new_from_template (gname, session->send_rtp_sink, templ);
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gst_ghost_pad_new_from_template (gname, session->send_rtp_src, templ);
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gst_pad_set_active (srcghost, TRUE);
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gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), srcghost);
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g_free (gname);
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@ -962,7 +968,7 @@ create_rtcp (GstRTPBin * rtpbin, GstPadTemplate * templ, const gchar * name)
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GstRTPBinSession *session;
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/* first get the session number */
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if (name == NULL || sscanf (name, "rtcp_src_%d", &sessid) != 1)
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if (name == NULL || sscanf (name, "send_rtcp_src_%d", &sessid) != 1)
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goto no_name;
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/* get or create session */
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@ -971,16 +977,17 @@ create_rtcp (GstRTPBin * rtpbin, GstPadTemplate * templ, const gchar * name)
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goto no_session;
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/* check if pad was requested */
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if (session->rtcp_src != NULL)
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if (session->send_rtcp_src != NULL)
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goto existed;
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/* get rtcp_src pad and store */
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session->rtcp_src =
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session->send_rtcp_src =
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gst_element_get_request_pad (session->session, "send_rtcp_src");
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if (session->rtcp_src == NULL)
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if (session->send_rtcp_src == NULL)
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goto pad_failed;
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result = gst_ghost_pad_new_from_template (name, session->rtcp_src, templ);
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result =
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gst_ghost_pad_new_from_template (name, session->send_rtcp_src, templ);
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gst_pad_set_active (result, TRUE);
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gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), result);
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@ -999,7 +1006,8 @@ no_session:
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}
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existed:
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{
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g_warning ("rtpbin: rtcp_src pad already requested for session %d", sessid);
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g_warning ("rtpbin: send_rtcp_src pad already requested for session %d",
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sessid);
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return NULL;
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}
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pad_failed:
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@ -1036,7 +1044,8 @@ gst_rtp_bin_request_new_pad (GstElement * element,
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} else if (templ == gst_element_class_get_pad_template (klass,
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"send_rtp_sink_%d")) {
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result = create_send_rtp (rtpbin, templ, name);
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} else if (templ == gst_element_class_get_pad_template (klass, "rtcp_src_%d")) {
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} else if (templ == gst_element_class_get_pad_template (klass,
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"send_rtcp_src_%d")) {
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result = create_rtcp (rtpbin, templ, name);
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} else
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goto wrong_template;
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@ -258,6 +258,8 @@ gst_rtp_pt_demux_chain (GstPad * pad, GstBuffer * buf)
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&ret);
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caps = g_value_get_boxed (&ret);
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if (caps == NULL)
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caps = GST_PAD_CAPS (rtpdemux->sink);
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if (!caps)
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goto no_caps;
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@ -451,6 +451,8 @@ gst_rtp_session_process_rtp (RTPSession * sess, RTPSource * src,
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rtpsession = GST_RTP_SESSION (user_data);
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priv = rtpsession->priv;
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GST_DEBUG_OBJECT (rtpsession, "reading receiving RTP packet");
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if (rtpsession->recv_rtp_src) {
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result = gst_pad_push (rtpsession->recv_rtp_src, buffer);
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} else {
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@ -473,6 +475,8 @@ gst_rtp_session_send_rtp (RTPSession * sess, RTPSource * src,
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rtpsession = GST_RTP_SESSION (user_data);
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priv = rtpsession->priv;
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GST_DEBUG_OBJECT (rtpsession, "sending RTP packet");
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if (rtpsession->send_rtp_src) {
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result = gst_pad_push (rtpsession->send_rtp_src, buffer);
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} else {
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@ -737,7 +741,7 @@ create_recv_rtp_sink (GstRTPSession * rtpsession)
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rtpsession->recv_rtp_sink =
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gst_pad_new_from_static_template (&rtpsession_recv_rtp_sink_template,
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NULL);
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"recv_rtp_sink");
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gst_pad_set_chain_function (rtpsession->recv_rtp_sink,
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gst_rtp_session_chain_recv_rtp);
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gst_pad_set_event_function (rtpsession->recv_rtp_sink,
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@ -766,7 +770,7 @@ create_recv_rtcp_sink (GstRTPSession * rtpsession)
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rtpsession->recv_rtcp_sink =
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gst_pad_new_from_static_template (&rtpsession_recv_rtcp_sink_template,
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NULL);
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"recv_rtcp_sink");
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gst_pad_set_chain_function (rtpsession->recv_rtcp_sink,
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gst_rtp_session_chain_recv_rtcp);
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gst_pad_set_event_function (rtpsession->recv_rtcp_sink,
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@ -795,18 +799,18 @@ create_send_rtp_sink (GstRTPSession * rtpsession)
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rtpsession->send_rtp_sink =
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gst_pad_new_from_static_template (&rtpsession_send_rtp_sink_template,
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NULL);
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"send_rtp_sink");
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gst_pad_set_chain_function (rtpsession->send_rtp_sink,
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gst_rtp_session_chain_send_rtp);
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gst_pad_set_event_function (rtpsession->send_rtp_sink,
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gst_rtp_session_event_send_rtp_sink);
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gst_pad_set_active (rtpsession->send_rtp_sink, TRUE);
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gst_element_add_pad (GST_ELEMENT_CAST (rtpsession),
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rtpsession->recv_rtcp_sink);
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rtpsession->send_rtp_sink);
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rtpsession->send_rtp_src =
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gst_pad_new_from_static_template (&rtpsession_send_rtp_src_template,
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NULL);
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"send_rtp_src");
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gst_pad_set_active (rtpsession->send_rtp_src, TRUE);
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gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->send_rtp_src);
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@ -824,7 +828,7 @@ create_send_rtcp_src (GstRTPSession * rtpsession)
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rtpsession->send_rtcp_src =
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gst_pad_new_from_static_template (&rtpsession_send_rtcp_src_template,
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NULL);
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"send_rtcp_src");
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gst_pad_set_active (rtpsession->send_rtcp_src, TRUE);
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gst_element_add_pad (GST_ELEMENT_CAST (rtpsession),
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rtpsession->send_rtcp_src);
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@ -622,6 +622,7 @@ source_push_rtp (RTPSource * source, GstBuffer * buffer, RTPSession * session)
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if (source == session->source) {
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GST_DEBUG ("source %08x pushed sender RTP packet", source->ssrc);
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RTP_SESSION_UNLOCK (session);
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if (session->callbacks.send_rtp)
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result =
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@ -629,8 +630,11 @@ source_push_rtp (RTPSource * source, GstBuffer * buffer, RTPSession * session)
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session->user_data);
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else
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gst_buffer_unref (buffer);
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} else {
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GST_DEBUG ("source %08x pushed receiver RTP packet", source->ssrc);
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RTP_SESSION_UNLOCK (session);
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if (session->callbacks.process_rtp)
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result =
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session->callbacks.process_rtp (session, source, buffer,
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@ -638,6 +642,8 @@ source_push_rtp (RTPSource * source, GstBuffer * buffer, RTPSession * session)
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else
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gst_buffer_unref (buffer);
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}
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RTP_SESSION_LOCK (session);
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return result;
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}
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@ -877,6 +883,7 @@ rtp_session_create_source (RTPSession * sess)
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}
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source = rtp_source_new (ssrc);
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g_object_ref (source);
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rtp_source_set_callbacks (source, &callbacks, sess);
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g_hash_table_insert (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc),
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source);
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/* we have one more source now */
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@ -1080,6 +1087,8 @@ rtp_session_process_sr (RTPSession * sess, GstRTCPPacket * packet,
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gst_rtcp_packet_get_rb (packet, i, &ssrc, &fractionlost,
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&packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
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GST_DEBUG ("RB %d: %08x, %u", i, ssrc, jitter);
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if (ssrc == sess->source->ssrc) {
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/* only deal with report blocks for our session, we update the stats of
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* the sender of the RTCP message. We could also compare our stats against
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@ -1361,7 +1370,8 @@ ignore:
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* @sess: an #RTPSession
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* @buffer: an RTP buffer
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*
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* Send the RTP buffer in the session manager.
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* Send the RTP buffer in the session manager. This function takes ownership of
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* @buffer.
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*
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* Returns: a #GstFlowReturn.
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*/
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@ -1375,9 +1385,19 @@ rtp_session_send_rtp (RTPSession * sess, GstBuffer * buffer)
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g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
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g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
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if (!gst_rtp_buffer_validate (buffer))
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goto invalid_packet;
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GST_DEBUG ("received RTP packet for sending");
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RTP_SESSION_LOCK (sess);
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source = sess->source;
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/* update last activity */
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if (sess->callbacks.get_time)
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source->last_rtp_activity =
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sess->callbacks.get_time (sess, sess->user_data);
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prevsender = RTP_SOURCE_IS_SENDER (source);
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/* we use our own source to send */
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|
|
@ -1388,6 +1408,14 @@ rtp_session_send_rtp (RTPSession * sess, GstBuffer * buffer)
|
|||
RTP_SESSION_UNLOCK (sess);
|
||||
|
||||
return result;
|
||||
|
||||
/* ERRORS */
|
||||
invalid_packet:
|
||||
{
|
||||
gst_buffer_unref (buffer);
|
||||
GST_DEBUG ("invalid RTP packet received");
|
||||
return GST_FLOW_OK;
|
||||
}
|
||||
}
|
||||
|
||||
static GstClockTime
|
||||
|
|
@ -1534,13 +1562,22 @@ session_start_rtcp (RTPSession * sess, ReportData * data)
|
|||
data->rtcp = gst_rtcp_buffer_new (sess->mtu);
|
||||
|
||||
if (RTP_SOURCE_IS_SENDER (own)) {
|
||||
guint64 ntptime;
|
||||
guint32 rtptime;
|
||||
|
||||
/* we are a sender, create SR */
|
||||
GST_DEBUG ("create SR for SSRC %08x", own->ssrc);
|
||||
gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_SR, packet);
|
||||
|
||||
/* fill in sender report info, FIXME NTP and RTP timestamps missing */
|
||||
/* convert clock time to NTP time */
|
||||
ntptime = gst_util_uint64_scale (data->time, (1LL << 32), GST_SECOND);
|
||||
ntptime += (2208988800LL << 32);
|
||||
|
||||
rtptime = 0;
|
||||
|
||||
/* fill in sender report info, FIXME RTP timestamps missing */
|
||||
gst_rtcp_packet_sr_set_sender_info (packet, own->ssrc,
|
||||
0, 0, own->stats.packets_sent, own->stats.octets_sent);
|
||||
ntptime, rtptime, own->stats.packets_sent, own->stats.octets_sent);
|
||||
} else {
|
||||
/* we are only receiver, create RR */
|
||||
GST_DEBUG ("create RR for SSRC %08x", own->ssrc);
|
||||
|
|
|
|||
|
|
@ -110,7 +110,7 @@ typedef GstClockTime (*RTPSessionGetTime) (RTPSession *sess, gpointer user_data)
|
|||
* @sess: an #RTPSession
|
||||
* @user_data: user data specified when registering
|
||||
*
|
||||
* This callback will be called when @sess needs to cancel the previous timeout.
|
||||
* This callback will be called when @sess needs to cancel the current timeout.
|
||||
* The currently running timeout should be canceled and a new reporting interval
|
||||
* should be requested from @sess.
|
||||
*/
|
||||
|
|
@ -122,6 +122,7 @@ typedef void (*RTPSessionReconsider) (RTPSession *sess, gpointer user_data);
|
|||
* @RTPSessionSendRTP: callback for sending RTP packets
|
||||
* @RTPSessionSendRTCP: callback for sending RTCP packets
|
||||
* @RTPSessionGetTime: callback for returning the current time
|
||||
* @RTPSessionReconsider: callback for reconsidering the timeout
|
||||
*
|
||||
* These callbacks can be installed on the session manager to get notification
|
||||
* when RTP and RTCP packets are ready for further processing. These callbacks
|
||||
|
|
|
|||
|
|
@ -453,18 +453,29 @@ GstFlowReturn
|
|||
rtp_source_send_rtp (RTPSource * src, GstBuffer * buffer)
|
||||
{
|
||||
GstFlowReturn result = GST_FLOW_OK;
|
||||
guint len;
|
||||
|
||||
g_return_val_if_fail (RTP_IS_SOURCE (src), GST_FLOW_ERROR);
|
||||
g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
|
||||
|
||||
len = gst_rtp_buffer_get_payload_len (buffer);
|
||||
|
||||
/* we are a sender now */
|
||||
src->is_sender = TRUE;
|
||||
|
||||
/* update stats for the SR */
|
||||
src->stats.packets_sent++;
|
||||
src->stats.octets_sent += len;
|
||||
|
||||
|
||||
/* push packet */
|
||||
if (src->callbacks.push_rtp)
|
||||
if (src->callbacks.push_rtp) {
|
||||
GST_DEBUG ("pushing RTP packet %u", src->stats.packets_sent);
|
||||
result = src->callbacks.push_rtp (src, buffer, src->user_data);
|
||||
else
|
||||
} else {
|
||||
GST_DEBUG ("no callback installed");
|
||||
gst_buffer_unref (buffer);
|
||||
}
|
||||
|
||||
return result;
|
||||
}
|
||||
|
|
|
|||
Loading…
Reference in a new issue