/* * Farsight Voice+Video library * * Copyright 2007 Collabora Ltd, * Copyright 2007 Nokia Corporation * @author: Philippe Kalaf . * Copyright 2007 Wim Taymans * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. * */ /** * SECTION:element-gstrtpjitterbuffer * * This element reorders and removes duplicate RTP packets as they are received * from a network source. It will also wait for missing packets up to a * configurable time limit using the #GstRtpJitterBuffer:latency property. * Packets arriving too late are considered to be lost packets. * * This element acts as a live element and so adds #GstRtpJitterBuffer:latency * to the pipeline. * * The element needs the clock-rate of the RTP payload in order to estimate the * delay. This information is obtained either from the caps on the sink pad or, * when no caps are present, from the #GstRtpJitterBuffer::request-pt-map signal. * To clear the previous pt-map use the #GstRtpJitterBuffer::clear-pt-map signal. * * This element will automatically be used inside gstrtpbin. * * * Example pipelines * |[ * gst-launch rtspsrc location=rtsp://192.168.1.133:8554/mpeg1or2AudioVideoTest ! gstrtpjitterbuffer ! rtpmpvdepay ! mpeg2dec ! xvimagesink * ]| Connect to a streaming server and decode the MPEG video. The jitterbuffer is * inserted into the pipeline to smooth out network jitter and to reorder the * out-of-order RTP packets. * * * Last reviewed on 2007-05-28 (0.10.5) */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include #include #include #include "gstrtpbin-marshal.h" #include "gstrtpjitterbuffer.h" #include "rtpjitterbuffer.h" #include "rtpstats.h" GST_DEBUG_CATEGORY (rtpjitterbuffer_debug); #define GST_CAT_DEFAULT (rtpjitterbuffer_debug) /* RTPJitterBuffer signals and args */ enum { SIGNAL_REQUEST_PT_MAP, SIGNAL_CLEAR_PT_MAP, SIGNAL_HANDLE_SYNC, SIGNAL_ON_NPT_STOP, SIGNAL_SET_ACTIVE, LAST_SIGNAL }; #define DEFAULT_LATENCY_MS 200 #define DEFAULT_DROP_ON_LATENCY FALSE #define DEFAULT_TS_OFFSET 0 #define DEFAULT_DO_LOST FALSE #define DEFAULT_MODE RTP_JITTER_BUFFER_MODE_SLAVE #define DEFAULT_PERCENT 0 enum { PROP_0, PROP_LATENCY, PROP_DROP_ON_LATENCY, PROP_TS_OFFSET, PROP_DO_LOST, PROP_MODE, PROP_PERCENT, PROP_LAST }; #define JBUF_LOCK(priv) (g_mutex_lock ((priv)->jbuf_lock)) #define JBUF_LOCK_CHECK(priv,label) G_STMT_START { \ JBUF_LOCK (priv); \ if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \ goto label; \ } G_STMT_END #define JBUF_UNLOCK(priv) (g_mutex_unlock ((priv)->jbuf_lock)) #define JBUF_WAIT(priv) (g_cond_wait ((priv)->jbuf_cond, (priv)->jbuf_lock)) #define JBUF_WAIT_CHECK(priv,label) G_STMT_START { \ JBUF_WAIT(priv); \ if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \ goto label; \ } G_STMT_END #define JBUF_SIGNAL(priv) (g_cond_signal ((priv)->jbuf_cond)) struct _GstRtpJitterBufferPrivate { GstPad *sinkpad, *srcpad; GstPad *rtcpsinkpad; RTPJitterBuffer *jbuf; GMutex *jbuf_lock; GCond *jbuf_cond; gboolean waiting; gboolean discont; gboolean active; guint64 out_offset; /* properties */ guint latency_ms; guint64 latency_ns; gboolean drop_on_latency; gint64 ts_offset; gboolean do_lost; /* the last seqnum we pushed out */ guint32 last_popped_seqnum; /* the next expected seqnum we push */ guint32 next_seqnum; /* last output time */ GstClockTime last_out_time; /* the next expected seqnum we receive */ guint32 next_in_seqnum; /* start and stop ranges */ GstClockTime npt_start; GstClockTime npt_stop; guint64 ext_timestamp; guint64 last_elapsed; guint64 estimated_eos; GstClockID eos_id; gboolean reached_npt_stop; /* state */ gboolean eos; /* clock rate and rtp timestamp offset */ gint last_pt; gint32 clock_rate; gint64 clock_base; gint64 prev_ts_offset; /* when we are shutting down */ GstFlowReturn srcresult; gboolean blocked; /* for sync */ GstSegment segment; GstClockID clock_id; gboolean unscheduled; /* the latency of the upstream peer, we have to take this into account when * synchronizing the buffers. */ GstClockTime peer_latency; /* some accounting */ guint64 num_late; guint64 num_duplicates; }; #define GST_RTP_JITTER_BUFFER_GET_PRIVATE(o) \ (G_TYPE_INSTANCE_GET_PRIVATE ((o), GST_TYPE_RTP_JITTER_BUFFER, \ GstRtpJitterBufferPrivate)) static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("application/x-rtp, " "clock-rate = (int) [ 1, 2147483647 ]" /* "payload = (int) , " * "encoding-name = (string) " */ ) ); static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_rtcp_template = GST_STATIC_PAD_TEMPLATE ("sink_rtcp", GST_PAD_SINK, GST_PAD_REQUEST, GST_STATIC_CAPS ("application/x-rtcp") ); static GstStaticPadTemplate gst_rtp_jitter_buffer_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("application/x-rtp" /* "payload = (int) , " * "clock-rate = (int) , " * "encoding-name = (string) " */ ) ); static guint gst_rtp_jitter_buffer_signals[LAST_SIGNAL] = { 0 }; GST_BOILERPLATE (GstRtpJitterBuffer, gst_rtp_jitter_buffer, GstElement, GST_TYPE_ELEMENT); /* object overrides */ static void gst_rtp_jitter_buffer_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_rtp_jitter_buffer_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static void gst_rtp_jitter_buffer_finalize (GObject * object); /* element overrides */ static GstStateChangeReturn gst_rtp_jitter_buffer_change_state (GstElement * element, GstStateChange transition); static GstPad *gst_rtp_jitter_buffer_request_new_pad (GstElement * element, GstPadTemplate * templ, const gchar * name); static void gst_rtp_jitter_buffer_release_pad (GstElement * element, GstPad * pad); static GstClock *gst_rtp_jitter_buffer_provide_clock (GstElement * element); /* pad overrides */ static GstCaps *gst_rtp_jitter_buffer_getcaps (GstPad * pad); static GstIterator *gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad); /* sinkpad overrides */ static gboolean gst_jitter_buffer_sink_setcaps (GstPad * pad, GstCaps * caps); static gboolean gst_rtp_jitter_buffer_sink_event (GstPad * pad, GstEvent * event); static GstFlowReturn gst_rtp_jitter_buffer_chain (GstPad * pad, GstBuffer * buffer); static gboolean gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad, GstEvent * event); static GstFlowReturn gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad, GstBuffer * buffer); /* srcpad overrides */ static gboolean gst_rtp_jitter_buffer_src_event (GstPad * pad, GstEvent * event); static gboolean gst_rtp_jitter_buffer_src_activate_push (GstPad * pad, gboolean active); static void gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer); static gboolean gst_rtp_jitter_buffer_query (GstPad * pad, GstQuery * query); static void gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer); static GstClockTime gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jitterbuffer, gboolean active, guint64 base_time); static void gst_rtp_jitter_buffer_base_init (gpointer klass) { GstElementClass *element_class = GST_ELEMENT_CLASS (klass); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_jitter_buffer_src_template)); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_jitter_buffer_sink_template)); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_jitter_buffer_sink_rtcp_template)); gst_element_class_set_details_simple (element_class, "RTP packet jitter-buffer", "Filter/Network/RTP", "A buffer that deals with network jitter and other transmission faults", "Philippe Kalaf , " "Wim Taymans "); } static void gst_rtp_jitter_buffer_class_init (GstRtpJitterBufferClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; g_type_class_add_private (klass, sizeof (GstRtpJitterBufferPrivate)); gobject_class->finalize = gst_rtp_jitter_buffer_finalize; gobject_class->set_property = gst_rtp_jitter_buffer_set_property; gobject_class->get_property = gst_rtp_jitter_buffer_get_property; /** * GstRtpJitterBuffer::latency: * * The maximum latency of the jitterbuffer. Packets will be kept in the buffer * for at most this time. */ g_object_class_install_property (gobject_class, PROP_LATENCY, g_param_spec_uint ("latency", "Buffer latency in ms", "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRtpJitterBuffer::drop-on-latency: * * Drop oldest buffers when the queue is completely filled. */ g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY, g_param_spec_boolean ("drop-on-latency", "Drop buffers when maximum latency is reached", "Tells the jitterbuffer to never exceed the given latency in size", DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRtpJitterBuffer::ts-offset: * * Adjust GStreamer output buffer timestamps in the jitterbuffer with offset. * This is mainly used to ensure interstream synchronisation. */ g_object_class_install_property (gobject_class, PROP_TS_OFFSET, g_param_spec_int64 ("ts-offset", "Timestamp Offset", "Adjust buffer timestamps with offset in nanoseconds", G_MININT64, G_MAXINT64, DEFAULT_TS_OFFSET, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRtpJitterBuffer::do-lost: * * Send out a GstRTPPacketLost event downstream when a packet is considered * lost. */ g_object_class_install_property (gobject_class, PROP_DO_LOST, g_param_spec_boolean ("do-lost", "Do Lost", "Send an event downstream when a packet is lost", DEFAULT_DO_LOST, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRtpJitterBuffer::mode: * * Control the buffering and timestamping mode used by the jitterbuffer. */ g_object_class_install_property (gobject_class, PROP_MODE, g_param_spec_enum ("mode", "Mode", "Control the buffering algorithm in use", RTP_TYPE_JITTER_BUFFER_MODE, DEFAULT_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRtpJitterBuffer::percent: * * The percent of the jitterbuffer that is filled. * * Since: 0.10.19 */ g_object_class_install_property (gobject_class, PROP_PERCENT, g_param_spec_int ("percent", "percent", "The buffer filled percent", 0, 100, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); /** * GstRtpJitterBuffer::request-pt-map: * @buffer: the object which received the signal * @pt: the pt * * Request the payload type as #GstCaps for @pt. */ gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP] = g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass, request_pt_map), NULL, NULL, gst_rtp_bin_marshal_BOXED__UINT, GST_TYPE_CAPS, 1, G_TYPE_UINT); /** * GstRtpJitterBuffer::handle-sync: * @buffer: the object which received the signal * @struct: a GstStructure containing sync values. * * Be notified of new sync values. */ gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC] = g_signal_new ("handle-sync", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass, handle_sync), NULL, NULL, g_cclosure_marshal_VOID__BOXED, G_TYPE_NONE, 1, GST_TYPE_STRUCTURE | G_SIGNAL_TYPE_STATIC_SCOPE); /** * GstRtpJitterBuffer::on-npt-stop * @buffer: the object which received the signal * * Signal that the jitterbufer has pushed the RTP packet that corresponds to * the npt-stop position. */ gst_rtp_jitter_buffer_signals[SIGNAL_ON_NPT_STOP] = g_signal_new ("on-npt-stop", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass, on_npt_stop), NULL, NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE); /** * GstRtpJitterBuffer::clear-pt-map: * @buffer: the object which received the signal * * Invalidate the clock-rate as obtained with the * #GstRtpJitterBuffer::request-pt-map signal. */ gst_rtp_jitter_buffer_signals[SIGNAL_CLEAR_PT_MAP] = g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpJitterBufferClass, clear_pt_map), NULL, NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE); /** * GstRtpJitterBuffer::set-active: * @buffer: the object which received the signal * * Start pushing out packets with the given base time. This signal is only * useful in buffering mode. * * Returns: the time of the last pushed packet. * * Since: 0.10.19 */ gst_rtp_jitter_buffer_signals[SIGNAL_SET_ACTIVE] = g_signal_new ("set-active", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpJitterBufferClass, set_active), NULL, NULL, gst_rtp_bin_marshal_UINT64__BOOL_UINT64, G_TYPE_UINT64, 2, G_TYPE_BOOLEAN, G_TYPE_UINT64); gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_change_state); gstelement_class->request_new_pad = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_request_new_pad); gstelement_class->release_pad = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_release_pad); gstelement_class->provide_clock = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_provide_clock); klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_clear_pt_map); klass->set_active = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_set_active); GST_DEBUG_CATEGORY_INIT (rtpjitterbuffer_debug, "gstrtpjitterbuffer", 0, "RTP Jitter Buffer"); } static void gst_rtp_jitter_buffer_init (GstRtpJitterBuffer * jitterbuffer, GstRtpJitterBufferClass * klass) { GstRtpJitterBufferPrivate *priv; priv = GST_RTP_JITTER_BUFFER_GET_PRIVATE (jitterbuffer); jitterbuffer->priv = priv; priv->latency_ms = DEFAULT_LATENCY_MS; priv->latency_ns = priv->latency_ms * GST_MSECOND; priv->drop_on_latency = DEFAULT_DROP_ON_LATENCY; priv->do_lost = DEFAULT_DO_LOST; priv->jbuf = rtp_jitter_buffer_new (); priv->jbuf_lock = g_mutex_new (); priv->jbuf_cond = g_cond_new (); /* reset skew detection initialy */ rtp_jitter_buffer_reset_skew (priv->jbuf); rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns); rtp_jitter_buffer_set_buffering (priv->jbuf, FALSE); priv->active = TRUE; priv->srcpad = gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_src_template, "src"); gst_pad_set_activatepush_function (priv->srcpad, GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_activate_push)); gst_pad_set_query_function (priv->srcpad, GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_query)); gst_pad_set_getcaps_function (priv->srcpad, GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_getcaps)); gst_pad_set_event_function (priv->srcpad, GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_event)); priv->sinkpad = gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_sink_template, "sink"); gst_pad_set_chain_function (priv->sinkpad, GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_chain)); gst_pad_set_event_function (priv->sinkpad, GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_event)); gst_pad_set_setcaps_function (priv->sinkpad, GST_DEBUG_FUNCPTR (gst_jitter_buffer_sink_setcaps)); gst_pad_set_getcaps_function (priv->sinkpad, GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_getcaps)); gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->srcpad); gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->sinkpad); } static void gst_rtp_jitter_buffer_finalize (GObject * object) { GstRtpJitterBuffer *jitterbuffer; jitterbuffer = GST_RTP_JITTER_BUFFER (object); g_mutex_free (jitterbuffer->priv->jbuf_lock); g_cond_free (jitterbuffer->priv->jbuf_cond); g_object_unref (jitterbuffer->priv->jbuf); G_OBJECT_CLASS (parent_class)->finalize (object); } static GstIterator * gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad) { GstRtpJitterBuffer *jitterbuffer; GstPad *otherpad = NULL; GstIterator *it; jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad)); if (pad == jitterbuffer->priv->sinkpad) { otherpad = jitterbuffer->priv->srcpad; } else if (pad == jitterbuffer->priv->srcpad) { otherpad = jitterbuffer->priv->sinkpad; } else if (pad == jitterbuffer->priv->rtcpsinkpad) { otherpad = NULL; } it = gst_iterator_new_single (GST_TYPE_PAD, otherpad, (GstCopyFunction) gst_object_ref, (GFreeFunc) gst_object_unref); gst_object_unref (jitterbuffer); return it; } static GstPad * create_rtcp_sink (GstRtpJitterBuffer * jitterbuffer) { GstRtpJitterBufferPrivate *priv; priv = jitterbuffer->priv; GST_DEBUG_OBJECT (jitterbuffer, "creating RTCP sink pad"); priv->rtcpsinkpad = gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_sink_rtcp_template, "sink_rtcp"); gst_pad_set_chain_function (priv->rtcpsinkpad, gst_rtp_jitter_buffer_chain_rtcp); gst_pad_set_event_function (priv->rtcpsinkpad, (GstPadEventFunction) gst_rtp_jitter_buffer_sink_rtcp_event); gst_pad_set_iterate_internal_links_function (priv->rtcpsinkpad, gst_rtp_jitter_buffer_iterate_internal_links); gst_pad_set_active (priv->rtcpsinkpad, TRUE); gst_element_add_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad); return priv->rtcpsinkpad; } static void remove_rtcp_sink (GstRtpJitterBuffer * jitterbuffer) { GstRtpJitterBufferPrivate *priv; priv = jitterbuffer->priv; GST_DEBUG_OBJECT (jitterbuffer, "removing RTCP sink pad"); gst_pad_set_active (priv->rtcpsinkpad, FALSE); gst_element_remove_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad); priv->rtcpsinkpad = NULL; } static GstPad * gst_rtp_jitter_buffer_request_new_pad (GstElement * element, GstPadTemplate * templ, const gchar * name) { GstRtpJitterBuffer *jitterbuffer; GstElementClass *klass; GstPad *result; GstRtpJitterBufferPrivate *priv; g_return_val_if_fail (templ != NULL, NULL); g_return_val_if_fail (GST_IS_RTP_JITTER_BUFFER (element), NULL); jitterbuffer = GST_RTP_JITTER_BUFFER (element); priv = jitterbuffer->priv; klass = GST_ELEMENT_GET_CLASS (element); GST_DEBUG_OBJECT (element, "requesting pad %s", GST_STR_NULL (name)); /* figure out the template */ if (templ == gst_element_class_get_pad_template (klass, "sink_rtcp")) { if (priv->rtcpsinkpad != NULL) goto exists; result = create_rtcp_sink (jitterbuffer); } else goto wrong_template; return result; /* ERRORS */ wrong_template: { g_warning ("gstrtpjitterbuffer: this is not our template"); return NULL; } exists: { g_warning ("gstrtpjitterbuffer: pad already requested"); return NULL; } } static void gst_rtp_jitter_buffer_release_pad (GstElement * element, GstPad * pad) { GstRtpJitterBuffer *jitterbuffer; GstRtpJitterBufferPrivate *priv; g_return_if_fail (GST_IS_RTP_JITTER_BUFFER (element)); g_return_if_fail (GST_IS_PAD (pad)); jitterbuffer = GST_RTP_JITTER_BUFFER (element); priv = jitterbuffer->priv; GST_DEBUG_OBJECT (element, "releasing pad %s:%s", GST_DEBUG_PAD_NAME (pad)); if (priv->rtcpsinkpad == pad) { remove_rtcp_sink (jitterbuffer); } else goto wrong_pad; return; /* ERRORS */ wrong_pad: { g_warning ("gstjitterbuffer: asked to release an unknown pad"); return; } } static GstClock * gst_rtp_jitter_buffer_provide_clock (GstElement * element) { return gst_system_clock_obtain (); } static void gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer) { GstRtpJitterBufferPrivate *priv; priv = jitterbuffer->priv; /* this will trigger a new pt-map request signal, FIXME, do something better. */ JBUF_LOCK (priv); priv->clock_rate = -1; JBUF_UNLOCK (priv); } static GstClockTime gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jbuf, gboolean active, guint64 offset) { GstRtpJitterBufferPrivate *priv; GstClockTime last_out; GstBuffer *head; priv = jbuf->priv; JBUF_LOCK (priv); GST_DEBUG_OBJECT (jbuf, "setting active %d with offset %" GST_TIME_FORMAT, active, GST_TIME_ARGS (offset)); if (active != priv->active) { /* add the amount of time spent in paused to the output offset. All * outgoing buffers will have this offset applied to their timestamps in * order to make them arrive in time in the sink. */ priv->out_offset = offset; GST_DEBUG_OBJECT (jbuf, "out offset %" GST_TIME_FORMAT, GST_TIME_ARGS (priv->out_offset)); priv->active = active; JBUF_SIGNAL (priv); } if (!active) { rtp_jitter_buffer_set_buffering (priv->jbuf, TRUE); } if ((head = rtp_jitter_buffer_peek (priv->jbuf))) { /* head buffer timestamp and offset gives our output time */ last_out = GST_BUFFER_TIMESTAMP (head) + priv->ts_offset; } else { /* use last known time when the buffer is empty */ last_out = priv->last_out_time; } JBUF_UNLOCK (priv); return last_out; } static GstCaps * gst_rtp_jitter_buffer_getcaps (GstPad * pad) { GstRtpJitterBuffer *jitterbuffer; GstRtpJitterBufferPrivate *priv; GstPad *other; GstCaps *caps; const GstCaps *templ; jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad)); priv = jitterbuffer->priv; other = (pad == priv->srcpad ? priv->sinkpad : priv->srcpad); caps = gst_pad_peer_get_caps (other); templ = gst_pad_get_pad_template_caps (pad); if (caps == NULL) { GST_DEBUG_OBJECT (jitterbuffer, "copy template"); caps = gst_caps_copy (templ); } else { GstCaps *intersect; GST_DEBUG_OBJECT (jitterbuffer, "intersect with template"); intersect = gst_caps_intersect (caps, templ); gst_caps_unref (caps); caps = intersect; } gst_object_unref (jitterbuffer); return caps; } /* * Must be called with JBUF_LOCK held */ static gboolean gst_jitter_buffer_sink_parse_caps (GstRtpJitterBuffer * jitterbuffer, GstCaps * caps) { GstRtpJitterBufferPrivate *priv; GstStructure *caps_struct; guint val; GstClockTime tval; priv = jitterbuffer->priv; /* first parse the caps */ caps_struct = gst_caps_get_structure (caps, 0); GST_DEBUG_OBJECT (jitterbuffer, "got caps"); /* we need a clock-rate to convert the rtp timestamps to GStreamer time and to * measure the amount of data in the buffer */ if (!gst_structure_get_int (caps_struct, "clock-rate", &priv->clock_rate)) goto error; if (priv->clock_rate <= 0) goto wrong_rate; GST_DEBUG_OBJECT (jitterbuffer, "got clock-rate %d", priv->clock_rate); /* The clock base is the RTP timestamp corrsponding to the npt-start value. We * can use this to track the amount of time elapsed on the sender. */ if (gst_structure_get_uint (caps_struct, "clock-base", &val)) priv->clock_base = val; else priv->clock_base = -1; priv->ext_timestamp = priv->clock_base; GST_DEBUG_OBJECT (jitterbuffer, "got clock-base %" G_GINT64_FORMAT, priv->clock_base); if (gst_structure_get_uint (caps_struct, "seqnum-base", &val)) { /* first expected seqnum, only update when we didn't have a previous base. */ if (priv->next_in_seqnum == -1) priv->next_in_seqnum = val; if (priv->next_seqnum == -1) priv->next_seqnum = val; } GST_DEBUG_OBJECT (jitterbuffer, "got seqnum-base %d", priv->next_in_seqnum); /* the start and stop times. The seqnum-base corresponds to the start time. We * will keep track of the seqnums on the output and when we reach the one * corresponding to npt-stop, we emit the npt-stop-reached signal */ if (gst_structure_get_clock_time (caps_struct, "npt-start", &tval)) priv->npt_start = tval; else priv->npt_start = 0; if (gst_structure_get_clock_time (caps_struct, "npt-stop", &tval)) priv->npt_stop = tval; else priv->npt_stop = -1; GST_DEBUG_OBJECT (jitterbuffer, "npt start/stop: %" GST_TIME_FORMAT "-%" GST_TIME_FORMAT, GST_TIME_ARGS (priv->npt_start), GST_TIME_ARGS (priv->npt_stop)); return TRUE; /* ERRORS */ error: { GST_DEBUG_OBJECT (jitterbuffer, "No clock-rate in caps!"); return FALSE; } wrong_rate: { GST_DEBUG_OBJECT (jitterbuffer, "Invalid clock-rate %d", priv->clock_rate); return FALSE; } } static gboolean gst_jitter_buffer_sink_setcaps (GstPad * pad, GstCaps * caps) { GstRtpJitterBuffer *jitterbuffer; GstRtpJitterBufferPrivate *priv; gboolean res; jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad)); priv = jitterbuffer->priv; JBUF_LOCK (priv); res = gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps); JBUF_UNLOCK (priv); /* set same caps on srcpad on success */ if (res) gst_pad_set_caps (priv->srcpad, caps); gst_object_unref (jitterbuffer); return res; } static void gst_rtp_jitter_buffer_flush_start (GstRtpJitterBuffer * jitterbuffer) { GstRtpJitterBufferPrivate *priv; priv = jitterbuffer->priv; JBUF_LOCK (priv); /* mark ourselves as flushing */ priv->srcresult = GST_FLOW_WRONG_STATE; GST_DEBUG_OBJECT (jitterbuffer, "Disabling pop on queue"); /* this unblocks any waiting pops on the src pad task */ JBUF_SIGNAL (priv); /* unlock clock, we just unschedule, the entry will be released by the * locking streaming thread. */ if (priv->clock_id) { gst_clock_id_unschedule (priv->clock_id); priv->unscheduled = TRUE; } JBUF_UNLOCK (priv); } static void gst_rtp_jitter_buffer_flush_stop (GstRtpJitterBuffer * jitterbuffer) { GstRtpJitterBufferPrivate *priv; priv = jitterbuffer->priv; JBUF_LOCK (priv); GST_DEBUG_OBJECT (jitterbuffer, "Enabling pop on queue"); /* Mark as non flushing */ priv->srcresult = GST_FLOW_OK; gst_segment_init (&priv->segment, GST_FORMAT_TIME); priv->last_popped_seqnum = -1; priv->last_out_time = -1; priv->next_seqnum = -1; priv->next_in_seqnum = -1; priv->clock_rate = -1; priv->eos = FALSE; priv->estimated_eos = -1; priv->last_elapsed = 0; priv->reached_npt_stop = FALSE; priv->ext_timestamp = -1; GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer"); rtp_jitter_buffer_flush (priv->jbuf); rtp_jitter_buffer_reset_skew (priv->jbuf); JBUF_UNLOCK (priv); } static gboolean gst_rtp_jitter_buffer_src_activate_push (GstPad * pad, gboolean active) { gboolean result = TRUE; GstRtpJitterBuffer *jitterbuffer = NULL; jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad)); if (active) { /* allow data processing */ gst_rtp_jitter_buffer_flush_stop (jitterbuffer); /* start pushing out buffers */ GST_DEBUG_OBJECT (jitterbuffer, "Starting task on srcpad"); gst_pad_start_task (jitterbuffer->priv->srcpad, (GstTaskFunction) gst_rtp_jitter_buffer_loop, jitterbuffer); } else { /* make sure all data processing stops ASAP */ gst_rtp_jitter_buffer_flush_start (jitterbuffer); /* NOTE this will hardlock if the state change is called from the src pad * task thread because we will _join() the thread. */ GST_DEBUG_OBJECT (jitterbuffer, "Stopping task on srcpad"); result = gst_pad_stop_task (pad); } gst_object_unref (jitterbuffer); return result; } static GstStateChangeReturn gst_rtp_jitter_buffer_change_state (GstElement * element, GstStateChange transition) { GstRtpJitterBuffer *jitterbuffer; GstRtpJitterBufferPrivate *priv; GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS; jitterbuffer = GST_RTP_JITTER_BUFFER (element); priv = jitterbuffer->priv; switch (transition) { case GST_STATE_CHANGE_NULL_TO_READY: break; case GST_STATE_CHANGE_READY_TO_PAUSED: JBUF_LOCK (priv); /* reset negotiated values */ priv->clock_rate = -1; priv->clock_base = -1; priv->peer_latency = 0; priv->last_pt = -1; /* block until we go to PLAYING */ priv->blocked = TRUE; JBUF_UNLOCK (priv); break; case GST_STATE_CHANGE_PAUSED_TO_PLAYING: JBUF_LOCK (priv); /* unblock to allow streaming in PLAYING */ priv->blocked = FALSE; JBUF_SIGNAL (priv); JBUF_UNLOCK (priv); break; default: break; } ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); switch (transition) { case GST_STATE_CHANGE_READY_TO_PAUSED: /* we are a live element because we sync to the clock, which we can only * do in the PLAYING state */ if (ret != GST_STATE_CHANGE_FAILURE) ret = GST_STATE_CHANGE_NO_PREROLL; break; case GST_STATE_CHANGE_PLAYING_TO_PAUSED: JBUF_LOCK (priv); /* block to stop streaming when PAUSED */ priv->blocked = TRUE; JBUF_UNLOCK (priv); if (ret != GST_STATE_CHANGE_FAILURE) ret = GST_STATE_CHANGE_NO_PREROLL; break; case GST_STATE_CHANGE_PAUSED_TO_READY: break; case GST_STATE_CHANGE_READY_TO_NULL: break; default: break; } return ret; } static gboolean gst_rtp_jitter_buffer_src_event (GstPad * pad, GstEvent * event) { gboolean ret = TRUE; GstRtpJitterBuffer *jitterbuffer; GstRtpJitterBufferPrivate *priv; jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad)); if (G_UNLIKELY (jitterbuffer == NULL)) { gst_event_unref (event); return FALSE; } priv = jitterbuffer->priv; GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event)); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_LATENCY: { GstClockTime latency; gst_event_parse_latency (event, &latency); JBUF_LOCK (priv); /* adjust the overall buffer delay to the total pipeline latency in * buffering mode because if downstream consumes too fast (because of * large latency or queues, we would start rebuffering again. */ if (rtp_jitter_buffer_get_mode (priv->jbuf) == RTP_JITTER_BUFFER_MODE_BUFFER) { rtp_jitter_buffer_set_delay (priv->jbuf, latency); } JBUF_UNLOCK (priv); ret = gst_pad_push_event (priv->sinkpad, event); break; } default: ret = gst_pad_push_event (priv->sinkpad, event); break; } gst_object_unref (jitterbuffer); return ret; } static gboolean gst_rtp_jitter_buffer_sink_event (GstPad * pad, GstEvent * event) { gboolean ret = TRUE; GstRtpJitterBuffer *jitterbuffer; GstRtpJitterBufferPrivate *priv; jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad)); if (G_UNLIKELY (jitterbuffer == NULL)) { gst_event_unref (event); return FALSE; } priv = jitterbuffer->priv; GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event)); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_NEWSEGMENT: { GstFormat format; gdouble rate, arate; gint64 start, stop, time; gboolean update; gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format, &start, &stop, &time); /* we need time for now */ if (format != GST_FORMAT_TIME) goto newseg_wrong_format; GST_DEBUG_OBJECT (jitterbuffer, "newsegment: update %d, rate %g, arate %g, start %" GST_TIME_FORMAT ", stop %" GST_TIME_FORMAT ", time %" GST_TIME_FORMAT, update, rate, arate, GST_TIME_ARGS (start), GST_TIME_ARGS (stop), GST_TIME_ARGS (time)); /* now configure the values, we need these to time the release of the * buffers on the srcpad. */ gst_segment_set_newsegment_full (&priv->segment, update, rate, arate, format, start, stop, time); /* FIXME, push SEGMENT in the queue. Sorting order might be difficult. */ ret = gst_pad_push_event (priv->srcpad, event); break; } case GST_EVENT_FLUSH_START: gst_rtp_jitter_buffer_flush_start (jitterbuffer); ret = gst_pad_push_event (priv->srcpad, event); break; case GST_EVENT_FLUSH_STOP: ret = gst_pad_push_event (priv->srcpad, event); ret = gst_rtp_jitter_buffer_src_activate_push (priv->srcpad, TRUE); break; case GST_EVENT_EOS: { /* push EOS in queue. We always push it at the head */ JBUF_LOCK (priv); /* check for flushing, we need to discard the event and return FALSE when * we are flushing */ ret = priv->srcresult == GST_FLOW_OK; if (ret && !priv->eos) { GST_INFO_OBJECT (jitterbuffer, "queuing EOS"); priv->eos = TRUE; JBUF_SIGNAL (priv); } else if (priv->eos) { GST_DEBUG_OBJECT (jitterbuffer, "dropping EOS, we are already EOS"); } else { GST_DEBUG_OBJECT (jitterbuffer, "dropping EOS, reason %s", gst_flow_get_name (priv->srcresult)); } JBUF_UNLOCK (priv); gst_event_unref (event); break; } default: ret = gst_pad_push_event (priv->srcpad, event); break; } done: gst_object_unref (jitterbuffer); return ret; /* ERRORS */ newseg_wrong_format: { GST_DEBUG_OBJECT (jitterbuffer, "received non TIME newsegment"); ret = FALSE; gst_event_unref (event); goto done; } } static gboolean gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad, GstEvent * event) { GstRtpJitterBuffer *jitterbuffer; GstRtpJitterBufferPrivate *priv; jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad)); priv = jitterbuffer->priv; GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event)); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_FLUSH_START: break; case GST_EVENT_FLUSH_STOP: break; default: break; } gst_event_unref (event); gst_object_unref (jitterbuffer); return TRUE; } /* * Must be called with JBUF_LOCK held, will release the LOCK when emiting the * signal. The function returns GST_FLOW_ERROR when a parsing error happened and * GST_FLOW_WRONG_STATE when the element is shutting down. On success * GST_FLOW_OK is returned. */ static GstFlowReturn gst_rtp_jitter_buffer_get_clock_rate (GstRtpJitterBuffer * jitterbuffer, guint8 pt) { GValue ret = { 0 }; GValue args[2] = { {0}, {0} }; GstCaps *caps; gboolean res; g_value_init (&args[0], GST_TYPE_ELEMENT); g_value_set_object (&args[0], jitterbuffer); g_value_init (&args[1], G_TYPE_UINT); g_value_set_uint (&args[1], pt); g_value_init (&ret, GST_TYPE_CAPS); g_value_set_boxed (&ret, NULL); JBUF_UNLOCK (jitterbuffer->priv); g_signal_emitv (args, gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP], 0, &ret); JBUF_LOCK_CHECK (jitterbuffer->priv, out_flushing); g_value_unset (&args[0]); g_value_unset (&args[1]); caps = (GstCaps *) g_value_dup_boxed (&ret); g_value_unset (&ret); if (!caps) goto no_caps; res = gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps); gst_caps_unref (caps); if (G_UNLIKELY (!res)) goto parse_failed; return GST_FLOW_OK; /* ERRORS */ no_caps: { GST_DEBUG_OBJECT (jitterbuffer, "could not get caps"); return GST_FLOW_ERROR; } out_flushing: { GST_DEBUG_OBJECT (jitterbuffer, "we are flushing"); return GST_FLOW_WRONG_STATE; } parse_failed: { GST_DEBUG_OBJECT (jitterbuffer, "parse failed"); return GST_FLOW_ERROR; } } /* call with jbuf lock held */ static void check_buffering_percent (GstRtpJitterBuffer * jitterbuffer, gint * percent) { GstRtpJitterBufferPrivate *priv = jitterbuffer->priv; /* too short a stream, or too close to EOS will never really fill buffer */ if (*percent != -1 && priv->npt_stop != -1 && priv->npt_stop - priv->npt_start <= rtp_jitter_buffer_get_delay (priv->jbuf)) { GST_DEBUG_OBJECT (jitterbuffer, "short stream; faking full buffer"); rtp_jitter_buffer_set_buffering (priv->jbuf, FALSE); *percent = 100; } } static void post_buffering_percent (GstRtpJitterBuffer * jitterbuffer, gint percent) { GstMessage *message; /* Post a buffering message */ message = gst_message_new_buffering (GST_OBJECT_CAST (jitterbuffer), percent); gst_message_set_buffering_stats (message, GST_BUFFERING_LIVE, -1, -1, -1); gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), message); } static GstFlowReturn gst_rtp_jitter_buffer_chain (GstPad * pad, GstBuffer * buffer) { GstRtpJitterBuffer *jitterbuffer; GstRtpJitterBufferPrivate *priv; guint16 seqnum; GstFlowReturn ret = GST_FLOW_OK; GstClockTime timestamp; guint64 latency_ts; gboolean tail; gint percent = -1; guint8 pt; jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad)); if (G_UNLIKELY (!gst_rtp_buffer_validate (buffer))) goto invalid_buffer; priv = jitterbuffer->priv; pt = gst_rtp_buffer_get_payload_type (buffer); /* take the timestamp of the buffer. This is the time when the packet was * received and is used to calculate jitter and clock skew. We will adjust * this timestamp with the smoothed value after processing it in the * jitterbuffer. */ timestamp = GST_BUFFER_TIMESTAMP (buffer); /* bring to running time */ timestamp = gst_segment_to_running_time (&priv->segment, GST_FORMAT_TIME, timestamp); seqnum = gst_rtp_buffer_get_seq (buffer); GST_DEBUG_OBJECT (jitterbuffer, "Received packet #%d at time %" GST_TIME_FORMAT, seqnum, GST_TIME_ARGS (timestamp)); JBUF_LOCK_CHECK (priv, out_flushing); if (G_UNLIKELY (priv->last_pt != pt)) { GstCaps *caps; GST_DEBUG_OBJECT (jitterbuffer, "pt changed from %u to %u", priv->last_pt, pt); priv->last_pt = pt; /* reset clock-rate so that we get a new one */ priv->clock_rate = -1; /* Try to get the clock-rate from the caps first if we can. If there are no * caps we must fire the signal to get the clock-rate. */ if ((caps = GST_BUFFER_CAPS (buffer))) { gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps); } } if (G_UNLIKELY (priv->clock_rate == -1)) { /* no clock rate given on the caps, try to get one with the signal */ if (gst_rtp_jitter_buffer_get_clock_rate (jitterbuffer, pt) == GST_FLOW_WRONG_STATE) goto out_flushing; if (G_UNLIKELY (priv->clock_rate == -1)) goto no_clock_rate; } /* don't accept more data on EOS */ if (G_UNLIKELY (priv->eos)) goto have_eos; /* now check against our expected seqnum */ if (G_LIKELY (priv->next_in_seqnum != -1)) { gint gap; gboolean reset = FALSE; gap = gst_rtp_buffer_compare_seqnum (priv->next_in_seqnum, seqnum); if (G_UNLIKELY (gap != 0)) { GST_DEBUG_OBJECT (jitterbuffer, "expected #%d, got #%d, gap of %d", priv->next_in_seqnum, seqnum, gap); /* priv->next_in_seqnum >= seqnum, this packet is too late or the * sender might have been restarted with different seqnum. */ if (gap < -RTP_MAX_MISORDER) { GST_DEBUG_OBJECT (jitterbuffer, "reset: buffer too old %d", gap); reset = TRUE; } /* priv->next_in_seqnum < seqnum, this is a new packet */ else if (G_UNLIKELY (gap > RTP_MAX_DROPOUT)) { GST_DEBUG_OBJECT (jitterbuffer, "reset: too many dropped packets %d", gap); reset = TRUE; } else { GST_DEBUG_OBJECT (jitterbuffer, "tolerable gap"); } } if (G_UNLIKELY (reset)) { GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer"); rtp_jitter_buffer_flush (priv->jbuf); rtp_jitter_buffer_reset_skew (priv->jbuf); priv->last_popped_seqnum = -1; priv->next_seqnum = seqnum; } } priv->next_in_seqnum = (seqnum + 1) & 0xffff; /* let's check if this buffer is too late, we can only accept packets with * bigger seqnum than the one we last pushed. */ if (G_LIKELY (priv->last_popped_seqnum != -1)) { gint gap; gap = gst_rtp_buffer_compare_seqnum (priv->last_popped_seqnum, seqnum); /* priv->last_popped_seqnum >= seqnum, we're too late. */ if (G_UNLIKELY (gap <= 0)) goto too_late; } /* let's drop oldest packet if the queue is already full and drop-on-latency * is set. We can only do this when there actually is a latency. When no * latency is set, we just pump it in the queue and let the other end push it * out as fast as possible. */ if (priv->latency_ms && priv->drop_on_latency) { latency_ts = gst_util_uint64_scale_int (priv->latency_ms, priv->clock_rate, 1000); if (G_UNLIKELY (rtp_jitter_buffer_get_ts_diff (priv->jbuf) >= latency_ts)) { GstBuffer *old_buf; old_buf = rtp_jitter_buffer_pop (priv->jbuf, &percent); GST_DEBUG_OBJECT (jitterbuffer, "Queue full, dropping old packet #%d", gst_rtp_buffer_get_seq (old_buf)); gst_buffer_unref (old_buf); } } /* we need to make the metadata writable before pushing it in the jitterbuffer * because the jitterbuffer will update the timestamp */ buffer = gst_buffer_make_metadata_writable (buffer); /* now insert the packet into the queue in sorted order. This function returns * FALSE if a packet with the same seqnum was already in the queue, meaning we * have a duplicate. */ if (G_UNLIKELY (!rtp_jitter_buffer_insert (priv->jbuf, buffer, timestamp, priv->clock_rate, &tail, &percent))) goto duplicate; /* signal addition of new buffer when the _loop is waiting. */ if (priv->waiting) JBUF_SIGNAL (priv); /* let's unschedule and unblock any waiting buffers. We only want to do this * when the tail buffer changed */ if (G_UNLIKELY (priv->clock_id && tail)) { GST_DEBUG_OBJECT (jitterbuffer, "Unscheduling waiting buffer, new tail buffer"); gst_clock_id_unschedule (priv->clock_id); priv->unscheduled = TRUE; } GST_DEBUG_OBJECT (jitterbuffer, "Pushed packet #%d, now %d packets, tail: %d", seqnum, rtp_jitter_buffer_num_packets (priv->jbuf), tail); check_buffering_percent (jitterbuffer, &percent); finished: JBUF_UNLOCK (priv); if (percent != -1) post_buffering_percent (jitterbuffer, percent); gst_object_unref (jitterbuffer); return ret; /* ERRORS */ invalid_buffer: { /* this is not fatal but should be filtered earlier */ GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL), ("Received invalid RTP payload, dropping")); gst_buffer_unref (buffer); gst_object_unref (jitterbuffer); return GST_FLOW_OK; } no_clock_rate: { GST_WARNING_OBJECT (jitterbuffer, "No clock-rate in caps!, dropping buffer"); gst_buffer_unref (buffer); goto finished; } out_flushing: { ret = priv->srcresult; GST_DEBUG_OBJECT (jitterbuffer, "flushing %s", gst_flow_get_name (ret)); gst_buffer_unref (buffer); goto finished; } have_eos: { ret = GST_FLOW_UNEXPECTED; GST_WARNING_OBJECT (jitterbuffer, "we are EOS, refusing buffer"); gst_buffer_unref (buffer); goto finished; } too_late: { GST_WARNING_OBJECT (jitterbuffer, "Packet #%d too late as #%d was already" " popped, dropping", seqnum, priv->last_popped_seqnum); priv->num_late++; gst_buffer_unref (buffer); goto finished; } duplicate: { GST_WARNING_OBJECT (jitterbuffer, "Duplicate packet #%d detected, dropping", seqnum); priv->num_duplicates++; gst_buffer_unref (buffer); goto finished; } } static GstClockTime apply_offset (GstRtpJitterBuffer * jitterbuffer, GstClockTime timestamp) { GstRtpJitterBufferPrivate *priv; priv = jitterbuffer->priv; if (timestamp == -1) return -1; /* apply the timestamp offset, this is used for inter stream sync */ timestamp += priv->ts_offset; /* add the offset, this is used when buffering */ timestamp += priv->out_offset; return timestamp; } static GstClockTime get_sync_time (GstRtpJitterBuffer * jitterbuffer, GstClockTime timestamp) { GstClockTime result; GstRtpJitterBufferPrivate *priv; priv = jitterbuffer->priv; result = timestamp + GST_ELEMENT_CAST (jitterbuffer)->base_time; /* add latency, this includes our own latency and the peer latency. */ result += priv->latency_ns; result += priv->peer_latency; return result; } static gboolean eos_reached (GstClock * clock, GstClockTime time, GstClockID id, GstRtpJitterBuffer * jitterbuffer) { GstRtpJitterBufferPrivate *priv; priv = jitterbuffer->priv; JBUF_LOCK_CHECK (priv, flushing); if (priv->waiting) { GST_INFO_OBJECT (jitterbuffer, "got the NPT timeout"); priv->reached_npt_stop = TRUE; JBUF_SIGNAL (priv); } JBUF_UNLOCK (priv); return TRUE; /* ERRORS */ flushing: { JBUF_UNLOCK (priv); return FALSE; } } static GstClockTime compute_elapsed (GstRtpJitterBuffer * jitterbuffer, GstBuffer * outbuf) { guint64 ext_time, elapsed; guint32 rtp_time; GstRtpJitterBufferPrivate *priv; priv = jitterbuffer->priv; rtp_time = gst_rtp_buffer_get_timestamp (outbuf); GST_LOG_OBJECT (jitterbuffer, "rtp %" G_GUINT32_FORMAT ", ext %" G_GUINT64_FORMAT, rtp_time, priv->ext_timestamp); if (rtp_time < priv->ext_timestamp) { ext_time = priv->ext_timestamp; } else { ext_time = gst_rtp_buffer_ext_timestamp (&priv->ext_timestamp, rtp_time); } if (ext_time > priv->clock_base) elapsed = ext_time - priv->clock_base; else elapsed = 0; elapsed = gst_util_uint64_scale_int (elapsed, GST_SECOND, priv->clock_rate); return elapsed; } /* * This funcion will push out buffers on the source pad. * * For each pushed buffer, the seqnum is recorded, if the next buffer B has a * different seqnum (missing packets before B), this function will wait for the * missing packet to arrive up to the timestamp of buffer B. */ static void gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer) { GstRtpJitterBufferPrivate *priv; GstBuffer *outbuf; GstFlowReturn result; guint16 seqnum; guint32 next_seqnum; GstClockTime timestamp, out_time; gboolean discont = FALSE; gint gap; GstClock *clock; GstClockID id; GstClockTime sync_time; gint percent = -1; priv = jitterbuffer->priv; JBUF_LOCK_CHECK (priv, flushing); again: GST_DEBUG_OBJECT (jitterbuffer, "Peeking item"); while (TRUE) { id = NULL; /* always wait if we are blocked */ if (G_LIKELY (!priv->blocked)) { /* we're buffering but not EOS, wait. */ if (!priv->eos && (!priv->active || rtp_jitter_buffer_is_buffering (priv->jbuf))) { GstClockTime elapsed, delay, left; if (priv->estimated_eos == -1) goto do_wait; outbuf = rtp_jitter_buffer_peek (priv->jbuf); if (outbuf != NULL) { elapsed = compute_elapsed (jitterbuffer, outbuf); if (GST_BUFFER_DURATION_IS_VALID (outbuf)) elapsed += GST_BUFFER_DURATION (outbuf); } else { GST_INFO_OBJECT (jitterbuffer, "no buffer, using last_elapsed"); elapsed = priv->last_elapsed; } delay = rtp_jitter_buffer_get_delay (priv->jbuf); if (priv->estimated_eos > elapsed) left = priv->estimated_eos - elapsed; else left = 0; GST_INFO_OBJECT (jitterbuffer, "buffering, elapsed %" GST_TIME_FORMAT " estimated_eos %" GST_TIME_FORMAT " left %" GST_TIME_FORMAT " delay %" GST_TIME_FORMAT, GST_TIME_ARGS (elapsed), GST_TIME_ARGS (priv->estimated_eos), GST_TIME_ARGS (left), GST_TIME_ARGS (delay)); if (left > delay) goto do_wait; } /* if we have a packet, we can exit the loop and grab it */ if (rtp_jitter_buffer_num_packets (priv->jbuf) > 0) break; /* no packets but we are EOS, do eos logic */ if (G_UNLIKELY (priv->eos)) goto do_eos; /* underrun, wait for packets or flushing now if we are expecting an EOS * timeout, set the async timer for it too */ if (priv->estimated_eos != -1 && !priv->reached_npt_stop) { sync_time = get_sync_time (jitterbuffer, priv->estimated_eos); GST_OBJECT_LOCK (jitterbuffer); clock = GST_ELEMENT_CLOCK (jitterbuffer); if (clock) { GST_INFO_OBJECT (jitterbuffer, "scheduling timeout"); id = gst_clock_new_single_shot_id (clock, sync_time); gst_clock_id_wait_async (id, (GstClockCallback) eos_reached, jitterbuffer); } GST_OBJECT_UNLOCK (jitterbuffer); } } do_wait: /* now we wait */ GST_DEBUG_OBJECT (jitterbuffer, "waiting"); priv->waiting = TRUE; JBUF_WAIT (priv); priv->waiting = FALSE; GST_DEBUG_OBJECT (jitterbuffer, "waiting done"); if (id) { /* unschedule any pending async notifications we might have */ gst_clock_id_unschedule (id); gst_clock_id_unref (id); } if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) goto flushing; if (id && priv->reached_npt_stop) { goto do_npt_stop; } } /* peek a buffer, we're just looking at the timestamp and the sequence number. * If all is fine, we'll pop and push it. If the sequence number is wrong we * wait on the timestamp. In the chain function we will unlock the wait when a * new buffer is available. The peeked buffer is valid for as long as we hold * the jitterbuffer lock. */ outbuf = rtp_jitter_buffer_peek (priv->jbuf); /* get the seqnum and the next expected seqnum */ seqnum = gst_rtp_buffer_get_seq (outbuf); next_seqnum = priv->next_seqnum; /* get the timestamp, this is already corrected for clock skew by the * jitterbuffer */ timestamp = GST_BUFFER_TIMESTAMP (outbuf); GST_DEBUG_OBJECT (jitterbuffer, "Peeked buffer #%d, expect #%d, timestamp %" GST_TIME_FORMAT ", now %d left", seqnum, next_seqnum, GST_TIME_ARGS (timestamp), rtp_jitter_buffer_num_packets (priv->jbuf)); /* apply our timestamp offset to the incomming buffer, this will be our output * timestamp. */ out_time = apply_offset (jitterbuffer, timestamp); /* get the gap between this and the previous packet. If we don't know the * previous packet seqnum assume no gap. */ if (G_LIKELY (next_seqnum != -1)) { gap = gst_rtp_buffer_compare_seqnum (next_seqnum, seqnum); /* if we have a packet that we already pushed or considered dropped, pop it * off and get the next packet */ if (G_UNLIKELY (gap < 0)) { GST_DEBUG_OBJECT (jitterbuffer, "Old packet #%d, next #%d dropping", seqnum, next_seqnum); outbuf = rtp_jitter_buffer_pop (priv->jbuf, &percent); gst_buffer_unref (outbuf); goto again; } } else { GST_DEBUG_OBJECT (jitterbuffer, "no next seqnum known, first packet"); gap = -1; } /* If we don't know what the next seqnum should be (== -1) we have to wait * because it might be possible that we are not receiving this buffer in-order, * a buffer with a lower seqnum could arrive later and we want to push that * earlier buffer before this buffer then. * If we know the expected seqnum, we can compare it to the current seqnum to * determine if we have missing a packet. If we have a missing packet (which * must be before this packet) we can wait for it until the deadline for this * packet expires. */ if (G_UNLIKELY (gap != 0 && out_time != -1)) { GstClockReturn ret; GstClockTime duration = GST_CLOCK_TIME_NONE; if (gap > 0) { /* we have a gap */ GST_DEBUG_OBJECT (jitterbuffer, "Sequence number GAP detected: expected %d instead of %d (%d missing)", next_seqnum, seqnum, gap); if (priv->last_out_time != -1) { GST_DEBUG_OBJECT (jitterbuffer, "out_time %" GST_TIME_FORMAT ", last %" GST_TIME_FORMAT, GST_TIME_ARGS (out_time), GST_TIME_ARGS (priv->last_out_time)); /* interpolate between the current time and the last time based on * number of packets we are missing, this is the estimated duration * for the missing packet based on equidistant packet spacing. Also make * sure we never go negative. */ if (out_time >= priv->last_out_time) duration = (out_time - priv->last_out_time) / (gap + 1); else goto lost; GST_DEBUG_OBJECT (jitterbuffer, "duration %" GST_TIME_FORMAT, GST_TIME_ARGS (duration)); /* add this duration to the timestamp of the last packet we pushed */ out_time = (priv->last_out_time + duration); } } else { /* we don't know what the next_seqnum should be, wait for the last * possible moment to push this buffer, maybe we get an earlier seqnum * while we wait */ GST_DEBUG_OBJECT (jitterbuffer, "First buffer %d, do sync", seqnum); } GST_OBJECT_LOCK (jitterbuffer); clock = GST_ELEMENT_CLOCK (jitterbuffer); if (!clock) { GST_OBJECT_UNLOCK (jitterbuffer); /* let's just push if there is no clock */ GST_DEBUG_OBJECT (jitterbuffer, "No clock, push right away"); goto push_buffer; } GST_DEBUG_OBJECT (jitterbuffer, "sync to timestamp %" GST_TIME_FORMAT, GST_TIME_ARGS (out_time)); /* prepare for sync against clock */ sync_time = get_sync_time (jitterbuffer, out_time); /* create an entry for the clock */ id = priv->clock_id = gst_clock_new_single_shot_id (clock, sync_time); priv->unscheduled = FALSE; GST_OBJECT_UNLOCK (jitterbuffer); /* release the lock so that the other end can push stuff or unlock */ JBUF_UNLOCK (priv); ret = gst_clock_id_wait (id, NULL); JBUF_LOCK (priv); /* and free the entry */ gst_clock_id_unref (id); priv->clock_id = NULL; /* at this point, the clock could have been unlocked by a timeout, a new * tail element was added to the queue or because we are shutting down. Check * for shutdown first. */ if G_UNLIKELY ((priv->srcresult != GST_FLOW_OK)) goto flushing; /* if we got unscheduled and we are not flushing, it's because a new tail * element became available in the queue or we flushed the queue. * Grab it and try to push or sync. */ if (ret == GST_CLOCK_UNSCHEDULED || priv->unscheduled) { GST_DEBUG_OBJECT (jitterbuffer, "Wait got unscheduled, will retry to push with new buffer"); goto again; } lost: /* we now timed out, this means we lost a packet or finished synchronizing * on the first buffer. */ if (gap > 0) { GstEvent *event; /* we had a gap and thus we lost a packet. Create an event for this. */ GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d lost", next_seqnum); priv->num_late++; discont = TRUE; /* update our expected next packet */ priv->last_popped_seqnum = next_seqnum; priv->last_out_time = out_time; priv->next_seqnum = (next_seqnum + 1) & 0xffff; if (priv->do_lost) { /* create paket lost event */ event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM, gst_structure_new ("GstRTPPacketLost", "seqnum", G_TYPE_UINT, (guint) next_seqnum, "timestamp", G_TYPE_UINT64, out_time, "duration", G_TYPE_UINT64, duration, NULL)); JBUF_UNLOCK (priv); gst_pad_push_event (priv->srcpad, event); JBUF_LOCK_CHECK (priv, flushing); } /* look for next packet */ goto again; } /* there was no known gap,just the first packet, exit the loop and push */ GST_DEBUG_OBJECT (jitterbuffer, "First packet #%d synced", seqnum); /* get new timestamp, latency might have changed */ out_time = apply_offset (jitterbuffer, timestamp); } push_buffer: /* when we get here we are ready to pop and push the buffer */ outbuf = rtp_jitter_buffer_pop (priv->jbuf, &percent); check_buffering_percent (jitterbuffer, &percent); if (G_UNLIKELY (discont || priv->discont)) { /* set DISCONT flag when we missed a packet. We pushed the buffer writable * into the jitterbuffer so we can modify now. */ GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT); priv->discont = FALSE; } /* apply timestamp with offset to buffer now */ GST_BUFFER_TIMESTAMP (outbuf) = out_time; /* update the elapsed time when we need to check against the npt stop time. */ if (priv->npt_stop != -1 && priv->ext_timestamp != -1 && priv->clock_base != -1 && priv->clock_rate > 0) { guint64 elapsed, estimated; elapsed = compute_elapsed (jitterbuffer, outbuf); if (elapsed > priv->last_elapsed || !priv->last_elapsed) { guint64 left; priv->last_elapsed = elapsed; left = priv->npt_stop - priv->npt_start; GST_LOG_OBJECT (jitterbuffer, "left %" GST_TIME_FORMAT, GST_TIME_ARGS (left)); if (elapsed > 0) estimated = gst_util_uint64_scale (out_time, left, elapsed); else { /* if there is almost nothing left, * we may never advance enough to end up in the above case */ if (left < GST_SECOND) estimated = GST_SECOND; else estimated = -1; } GST_LOG_OBJECT (jitterbuffer, "elapsed %" GST_TIME_FORMAT ", estimated %" GST_TIME_FORMAT, GST_TIME_ARGS (elapsed), GST_TIME_ARGS (estimated)); priv->estimated_eos = estimated; } } /* now we are ready to push the buffer. Save the seqnum and release the lock * so the other end can push stuff in the queue again. */ priv->last_popped_seqnum = seqnum; priv->last_out_time = out_time; priv->next_seqnum = (seqnum + 1) & 0xffff; JBUF_UNLOCK (priv); if (percent != -1) post_buffering_percent (jitterbuffer, percent); /* push buffer */ GST_DEBUG_OBJECT (jitterbuffer, "Pushing buffer %d, timestamp %" GST_TIME_FORMAT, seqnum, GST_TIME_ARGS (out_time)); result = gst_pad_push (priv->srcpad, outbuf); if (G_UNLIKELY (result != GST_FLOW_OK)) goto pause; return; /* ERRORS */ do_eos: { /* store result, we are flushing now */ GST_DEBUG_OBJECT (jitterbuffer, "We are EOS, pushing EOS downstream"); priv->srcresult = GST_FLOW_UNEXPECTED; gst_pad_pause_task (priv->srcpad); JBUF_UNLOCK (priv); gst_pad_push_event (priv->srcpad, gst_event_new_eos ()); return; } do_npt_stop: { /* store result, we are flushing now */ GST_DEBUG_OBJECT (jitterbuffer, "We reached the NPT stop"); JBUF_UNLOCK (priv); g_signal_emit (jitterbuffer, gst_rtp_jitter_buffer_signals[SIGNAL_ON_NPT_STOP], 0, NULL); return; } flushing: { GST_DEBUG_OBJECT (jitterbuffer, "we are flushing"); gst_pad_pause_task (priv->srcpad); JBUF_UNLOCK (priv); return; } pause: { GST_DEBUG_OBJECT (jitterbuffer, "pausing task, reason %s", gst_flow_get_name (result)); JBUF_LOCK (priv); /* store result */ priv->srcresult = result; /* we don't post errors or anything because upstream will do that for us * when we pass the return value upstream. */ gst_pad_pause_task (priv->srcpad); JBUF_UNLOCK (priv); return; } } static GstFlowReturn gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad, GstBuffer * buffer) { GstRtpJitterBuffer *jitterbuffer; GstRtpJitterBufferPrivate *priv; GstFlowReturn ret = GST_FLOW_OK; guint64 base_rtptime, base_time; guint32 clock_rate; guint64 last_rtptime; guint32 ssrc; GstRTCPPacket packet; guint64 ext_rtptime, diff; guint32 rtptime; gboolean drop = FALSE; jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad)); if (G_UNLIKELY (!gst_rtcp_buffer_validate (buffer))) goto invalid_buffer; priv = jitterbuffer->priv; if (!gst_rtcp_buffer_get_first_packet (buffer, &packet)) goto invalid_buffer; /* first packet must be SR or RR or else the validate would have failed */ switch (gst_rtcp_packet_get_type (&packet)) { case GST_RTCP_TYPE_SR: gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, NULL, &rtptime, NULL, NULL); break; default: goto ignore_buffer; } GST_DEBUG_OBJECT (jitterbuffer, "received RTCP of SSRC %08x", ssrc); JBUF_LOCK (priv); /* convert the RTP timestamp to our extended timestamp, using the same offset * we used in the jitterbuffer */ ext_rtptime = priv->jbuf->ext_rtptime; ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime); /* get the last values from the jitterbuffer */ rtp_jitter_buffer_get_sync (priv->jbuf, &base_rtptime, &base_time, &clock_rate, &last_rtptime); GST_DEBUG_OBJECT (jitterbuffer, "ext SR %" G_GUINT64_FORMAT ", base %" G_GUINT64_FORMAT ", clock-rate %" G_GUINT32_FORMAT, ext_rtptime, base_rtptime, clock_rate); if (base_rtptime == -1 || clock_rate == -1 || base_time == -1) { GST_DEBUG_OBJECT (jitterbuffer, "dropping, no RTP values"); drop = TRUE; } else { /* we can't accept anything that happened before we did the last resync */ if (base_rtptime > ext_rtptime) { GST_DEBUG_OBJECT (jitterbuffer, "dropping, older than base time"); drop = TRUE; } else { /* the SR RTP timestamp must be something close to what we last observed * in the jitterbuffer */ if (ext_rtptime > last_rtptime) { /* check how far ahead it is to our RTP timestamps */ diff = ext_rtptime - last_rtptime; /* if bigger than 1 second, we drop it */ if (diff > clock_rate) { GST_DEBUG_OBJECT (jitterbuffer, "dropping, too far ahead"); drop = TRUE; } GST_DEBUG_OBJECT (jitterbuffer, "ext last %" G_GUINT64_FORMAT ", diff %" G_GUINT64_FORMAT, last_rtptime, diff); } } } JBUF_UNLOCK (priv); if (!drop) { GstStructure *s; s = gst_structure_new ("application/x-rtp-sync", "base-rtptime", G_TYPE_UINT64, base_rtptime, "base-time", G_TYPE_UINT64, base_time, "clock-rate", G_TYPE_UINT, clock_rate, "sr-ext-rtptime", G_TYPE_UINT64, ext_rtptime, "sr-buffer", GST_TYPE_BUFFER, buffer, NULL); GST_DEBUG_OBJECT (jitterbuffer, "signaling sync"); g_signal_emit (jitterbuffer, gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC], 0, s); gst_structure_free (s); } else { GST_DEBUG_OBJECT (jitterbuffer, "dropping RTCP packet"); ret = GST_FLOW_OK; } done: gst_buffer_unref (buffer); gst_object_unref (jitterbuffer); return ret; invalid_buffer: { /* this is not fatal but should be filtered earlier */ GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL), ("Received invalid RTCP payload, dropping")); ret = GST_FLOW_OK; goto done; } ignore_buffer: { GST_DEBUG_OBJECT (jitterbuffer, "ignoring RTCP packet"); ret = GST_FLOW_OK; goto done; } } static gboolean gst_rtp_jitter_buffer_query (GstPad * pad, GstQuery * query) { GstRtpJitterBuffer *jitterbuffer; GstRtpJitterBufferPrivate *priv; gboolean res = FALSE; jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad)); if (G_UNLIKELY (jitterbuffer == NULL)) return FALSE; priv = jitterbuffer->priv; switch (GST_QUERY_TYPE (query)) { case GST_QUERY_LATENCY: { /* We need to send the query upstream and add the returned latency to our * own */ GstClockTime min_latency, max_latency; gboolean us_live; GstClockTime our_latency; if ((res = gst_pad_peer_query (priv->sinkpad, query))) { gst_query_parse_latency (query, &us_live, &min_latency, &max_latency); GST_DEBUG_OBJECT (jitterbuffer, "Peer latency: min %" GST_TIME_FORMAT " max %" GST_TIME_FORMAT, GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency)); /* store this so that we can safely sync on the peer buffers. */ JBUF_LOCK (priv); priv->peer_latency = min_latency; our_latency = priv->latency_ns; JBUF_UNLOCK (priv); GST_DEBUG_OBJECT (jitterbuffer, "Our latency: %" GST_TIME_FORMAT, GST_TIME_ARGS (our_latency)); /* we add some latency but can buffer an infinite amount of time */ min_latency += our_latency; max_latency = -1; GST_DEBUG_OBJECT (jitterbuffer, "Calculated total latency : min %" GST_TIME_FORMAT " max %" GST_TIME_FORMAT, GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency)); gst_query_set_latency (query, TRUE, min_latency, max_latency); } break; } case GST_QUERY_POSITION: { GstClockTime start, last_out; GstFormat fmt; gst_query_parse_position (query, &fmt, NULL); if (fmt != GST_FORMAT_TIME) { res = gst_pad_query_default (pad, query); break; } JBUF_LOCK (priv); start = priv->npt_start; last_out = priv->last_out_time; JBUF_UNLOCK (priv); GST_DEBUG_OBJECT (jitterbuffer, "npt start %" GST_TIME_FORMAT ", last out %" GST_TIME_FORMAT, GST_TIME_ARGS (start), GST_TIME_ARGS (last_out)); if (GST_CLOCK_TIME_IS_VALID (start) && GST_CLOCK_TIME_IS_VALID (last_out)) { /* bring 0-based outgoing time to stream time */ gst_query_set_position (query, GST_FORMAT_TIME, start + last_out); res = TRUE; } else { res = gst_pad_query_default (pad, query); } break; } default: res = gst_pad_query_default (pad, query); break; } gst_object_unref (jitterbuffer); return res; } static void gst_rtp_jitter_buffer_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstRtpJitterBuffer *jitterbuffer; GstRtpJitterBufferPrivate *priv; jitterbuffer = GST_RTP_JITTER_BUFFER (object); priv = jitterbuffer->priv; switch (prop_id) { case PROP_LATENCY: { guint new_latency, old_latency; new_latency = g_value_get_uint (value); JBUF_LOCK (priv); old_latency = priv->latency_ms; priv->latency_ms = new_latency; priv->latency_ns = priv->latency_ms * GST_MSECOND; rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns); JBUF_UNLOCK (priv); /* post message if latency changed, this will inform the parent pipeline * that a latency reconfiguration is possible/needed. */ if (new_latency != old_latency) { GST_DEBUG_OBJECT (jitterbuffer, "latency changed to: %" GST_TIME_FORMAT, GST_TIME_ARGS (new_latency * GST_MSECOND)); gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), gst_message_new_latency (GST_OBJECT_CAST (jitterbuffer))); } break; } case PROP_DROP_ON_LATENCY: JBUF_LOCK (priv); priv->drop_on_latency = g_value_get_boolean (value); JBUF_UNLOCK (priv); break; case PROP_TS_OFFSET: JBUF_LOCK (priv); priv->ts_offset = g_value_get_int64 (value); /* FIXME, we don't really have a method for signaling a timestamp * DISCONT without also making this a data discont. */ /* priv->discont = TRUE; */ JBUF_UNLOCK (priv); break; case PROP_DO_LOST: JBUF_LOCK (priv); priv->do_lost = g_value_get_boolean (value); JBUF_UNLOCK (priv); break; case PROP_MODE: JBUF_LOCK (priv); rtp_jitter_buffer_set_mode (priv->jbuf, g_value_get_enum (value)); JBUF_UNLOCK (priv); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_rtp_jitter_buffer_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstRtpJitterBuffer *jitterbuffer; GstRtpJitterBufferPrivate *priv; jitterbuffer = GST_RTP_JITTER_BUFFER (object); priv = jitterbuffer->priv; switch (prop_id) { case PROP_LATENCY: JBUF_LOCK (priv); g_value_set_uint (value, priv->latency_ms); JBUF_UNLOCK (priv); break; case PROP_DROP_ON_LATENCY: JBUF_LOCK (priv); g_value_set_boolean (value, priv->drop_on_latency); JBUF_UNLOCK (priv); break; case PROP_TS_OFFSET: JBUF_LOCK (priv); g_value_set_int64 (value, priv->ts_offset); JBUF_UNLOCK (priv); break; case PROP_DO_LOST: JBUF_LOCK (priv); g_value_set_boolean (value, priv->do_lost); JBUF_UNLOCK (priv); break; case PROP_MODE: JBUF_LOCK (priv); g_value_set_enum (value, rtp_jitter_buffer_get_mode (priv->jbuf)); JBUF_UNLOCK (priv); break; case PROP_PERCENT: { gint percent; JBUF_LOCK (priv); if (priv->srcresult != GST_FLOW_OK) percent = 100; else percent = rtp_jitter_buffer_get_percent (priv->jbuf); g_value_set_int (value, percent); JBUF_UNLOCK (priv); break; } default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } }