diff options
| -rw-r--r-- | gst/rtpmanager/gstrtpjitterbuffer.c | 6 | ||||
| -rw-r--r-- | gst/rtpmanager/gstrtpsession.c | 70 | ||||
| -rw-r--r-- | gst/rtpmanager/rtpsource.c | 3 |
3 files changed, 69 insertions, 10 deletions
diff --git a/gst/rtpmanager/gstrtpjitterbuffer.c b/gst/rtpmanager/gstrtpjitterbuffer.c index 2e44425a..f4d59bb6 100644 --- a/gst/rtpmanager/gstrtpjitterbuffer.c +++ b/gst/rtpmanager/gstrtpjitterbuffer.c @@ -942,6 +942,7 @@ again: if (priv->eos) goto do_eos; } + /* wait for packets or flushing now */ JBUF_WAIT_CHECK (priv, flushing); } @@ -1004,7 +1005,6 @@ again: running_time = gst_segment_to_running_time (&priv->segment, GST_FORMAT_TIME, timestamp); - /* correct for sync against the gstreamer clock, add latency */ GST_OBJECT_LOCK (jitterbuffer); clock = GST_ELEMENT_CLOCK (jitterbuffer); if (!clock) { @@ -1013,7 +1013,7 @@ again: goto push_buffer; } - /* add latency */ + /* add latency, this includes our own latency and the peer latency. */ running_time += (priv->latency_ms * GST_MSECOND); running_time += priv->peer_latency; @@ -1050,7 +1050,7 @@ again: if (ret == GST_CLOCK_UNSCHEDULED) { GST_DEBUG_OBJECT (jitterbuffer, "Wait got unscheduled, will retry to push with new buffer"); - /* reinserting popped buffer into queue */ + /* reinsert popped buffer into queue */ if (!rtp_jitter_buffer_insert (priv->jbuf, outbuf)) { GST_DEBUG_OBJECT (jitterbuffer, "Duplicate packet #%d detected, dropping", seqnum); diff --git a/gst/rtpmanager/gstrtpsession.c b/gst/rtpmanager/gstrtpsession.c index 554422fc..985a3713 100644 --- a/gst/rtpmanager/gstrtpsession.c +++ b/gst/rtpmanager/gstrtpsession.c @@ -227,10 +227,15 @@ struct _GstRtpSessionPrivate { GMutex *lock; RTPSession *session; + /* thread for sending out RTCP */ GstClockID id; gboolean stop_thread; GThread *thread; + + /* caps mapping */ + guint8 pt; + gint clock_rate; }; /* callbacks to handle actions from the session manager */ @@ -657,6 +662,7 @@ gst_rtp_session_change_state (GstElement * element, GstStateChange transition) case GST_STATE_CHANGE_NULL_TO_READY: break; case GST_STATE_CHANGE_READY_TO_PAUSED: + priv->clock_rate = -1; break; case GST_STATE_CHANGE_PAUSED_TO_PLAYING: break; @@ -778,6 +784,31 @@ gst_rtp_session_send_rtcp (RTPSession * sess, RTPSource * src, return result; } +static gboolean +gst_rtp_session_parse_caps (GstRtpSession * rtpsession, GstCaps * caps) +{ + GstRtpSessionPrivate *priv; + const GstStructure *caps_struct; + + priv = rtpsession->priv; + + GST_DEBUG_OBJECT (rtpsession, "parsing caps"); + + caps_struct = gst_caps_get_structure (caps, 0); + if (!gst_structure_get_int (caps_struct, "clock-rate", &priv->clock_rate)) + goto no_clock_rate; + + GST_DEBUG_OBJECT (rtpsession, "parsed clock-rate %d", priv->clock_rate); + + return TRUE; + + /* ERRORS */ +no_clock_rate: + { + GST_DEBUG_OBJECT (rtpsession, "No clock-rate in caps!"); + return FALSE; + } +} /* called when the session manager needs the clock rate */ static gint @@ -786,12 +817,17 @@ gst_rtp_session_clock_rate (RTPSession * sess, guint8 payload, { gint result = -1; GstRtpSession *rtpsession; + GstRtpSessionPrivate *priv; GValue ret = { 0 }; GValue args[2] = { {0}, {0} }; GstCaps *caps; - const GstStructure *caps_struct; rtpsession = GST_RTP_SESSION_CAST (user_data); + priv = rtpsession->priv; + + /* if we have it, return it */ + if (priv->clock_rate != -1) + return priv->clock_rate; g_value_init (&args[0], GST_TYPE_ELEMENT); g_value_set_object (&args[0], rtpsession); @@ -808,11 +844,10 @@ gst_rtp_session_clock_rate (RTPSession * sess, guint8 payload, if (!caps) goto no_caps; - caps_struct = gst_caps_get_structure (caps, 0); - if (!gst_structure_get_int (caps_struct, "clock-rate", &result)) - goto no_clock_rate; + if (!gst_rtp_session_parse_caps (rtpsession, caps)) + goto parse_failed; - GST_DEBUG_OBJECT (rtpsession, "parsed clock-rate %d", result); + result = priv->clock_rate; return result; @@ -822,9 +857,9 @@ no_caps: GST_DEBUG_OBJECT (rtpsession, "could not get caps"); return -1; } -no_clock_rate: +parse_failed: { - GST_DEBUG_OBJECT (rtpsession, "could not clock-rate from caps"); + GST_DEBUG_OBJECT (rtpsession, "failed to parse caps"); return -1; } } @@ -887,6 +922,23 @@ gst_rtp_session_event_recv_rtp_sink (GstPad * pad, GstEvent * event) return ret; } +static gboolean +gst_rtp_session_sink_setcaps (GstPad * pad, GstCaps * caps) +{ + gboolean res; + GstRtpSession *rtpsession; + GstRtpSessionPrivate *priv; + + rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad)); + priv = rtpsession->priv; + + res = gst_rtp_session_parse_caps (rtpsession, caps); + + gst_object_unref (rtpsession); + + return res; +} + /* receive a packet from a sender, send it to the RTP session manager and * forward the packet on the rtp_src pad */ @@ -1051,6 +1103,8 @@ create_recv_rtp_sink (GstRtpSession * rtpsession) gst_rtp_session_chain_recv_rtp); gst_pad_set_event_function (rtpsession->recv_rtp_sink, gst_rtp_session_event_recv_rtp_sink); + gst_pad_set_setcaps_function (rtpsession->recv_rtp_sink, + gst_rtp_session_sink_setcaps); gst_pad_set_active (rtpsession->recv_rtp_sink, TRUE); gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->recv_rtp_sink); @@ -1109,6 +1163,8 @@ create_send_rtp_sink (GstRtpSession * rtpsession) gst_rtp_session_chain_send_rtp); gst_pad_set_event_function (rtpsession->send_rtp_sink, gst_rtp_session_event_send_rtp_sink); + gst_pad_set_setcaps_function (rtpsession->send_rtp_sink, + gst_rtp_session_sink_setcaps); gst_pad_set_active (rtpsession->send_rtp_sink, TRUE); gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->send_rtp_sink); diff --git a/gst/rtpmanager/rtpsource.c b/gst/rtpmanager/rtpsource.c index ad491bd0..24bb8466 100644 --- a/gst/rtpmanager/rtpsource.c +++ b/gst/rtpmanager/rtpsource.c @@ -481,6 +481,9 @@ rtp_source_send_rtp (RTPSource * src, GstBuffer * buffer) if (timestamp != -1) src->last_timestamp = timestamp; + if (src->clock_rate == -1) + get_clock_rate (src, gst_rtp_buffer_get_payload_type (buffer)); + /* push packet */ if (src->callbacks.push_rtp) { guint32 ssrc; |