/* * Farsight Voice+Video library * * Copyright 2007 Collabora Ltd, * Copyright 2007 Nokia Corporation * @author: Philippe Kalaf . * Copyright 2007 Wim Taymans * Copyright 2015 Kurento (http://kurento.org/) * @author: Miguel París * Copyright 2016 Pexip AS * @author: Havard Graff * @author: Stian Selnes * * 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., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, USA. * */ /** * SECTION:element-rtpjitterbuffer * @title: rtpjitterbuffer * * This element reorders and removes duplicate RTP packets as they are received * from a network source. * * 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. * * The rtpjitterbuffer will 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. If the #GstRtpJitterBuffer:do-lost * property is set, lost packets will result in a custom serialized downstream * event of name GstRTPPacketLost. The lost packet events are usually used by a * depayloader or other element to create concealment data or some other logic * to gracefully handle the missing packets. * * The jitterbuffer will use the DTS (or PTS if no DTS is set) of the incoming * buffer and the rtptime inside the RTP packet to create a PTS on the outgoing * buffer. * * The jitterbuffer can also be configured to send early retransmission events * upstream by setting the #GstRtpJitterBuffer:do-retransmission property. In * this mode, the jitterbuffer tries to estimate when a packet should arrive and * sends a custom upstream event named GstRTPRetransmissionRequest when the * packet is considered late. The initial expected packet arrival time is * calculated as follows: * * - If seqnum N arrived at time T, seqnum N+1 is expected to arrive at * T + packet-spacing + #GstRtpJitterBuffer:rtx-delay. The packet spacing is * calculated from the DTS (or PTS is no DTS) of two consecutive RTP * packets with different rtptime. * * - If seqnum N0 arrived at time T0 and seqnum Nm arrived at time Tm, * seqnum Ni is expected at time Ti = T0 + i*(Tm - T0)/(Nm - N0). Any * previously scheduled timeout is overwritten. * * - If seqnum N arrived, all seqnum older than * N - #GstRtpJitterBuffer:rtx-delay-reorder are considered late * immediately. This is to request fast feedback for abnormally reorder * packets before any of the previous timeouts is triggered. * * A late packet triggers the GstRTPRetransmissionRequest custom upstream * event. After the initial timeout expires and the retransmission event is * sent, the timeout is scheduled for * T + #GstRtpJitterBuffer:rtx-retry-timeout. If the missing packet did not * arrive after #GstRtpJitterBuffer:rtx-retry-timeout, a new * GstRTPRetransmissionRequest is sent upstream and the timeout is rescheduled * again for T + #GstRtpJitterBuffer:rtx-retry-timeout. This repeats until * #GstRtpJitterBuffer:rtx-retry-period elapsed, at which point no further * retransmission requests are sent and the regular logic is performed to * schedule a lost packet as discussed above. * * This element acts as a live element and so adds #GstRtpJitterBuffer:latency * to the pipeline. * * This element will automatically be used inside rtpbin. * * ## Example pipelines * |[ * gst-launch-1.0 rtspsrc location=rtsp://192.168.1.133:8554/mpeg1or2AudioVideoTest ! rtpjitterbuffer ! 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. * */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include #include #include #include #include #include "gstrtpjitterbuffer.h" #include "rtpjitterbuffer.h" #include "rtpstats.h" #include "rtptimerqueue.h" #include 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_MAX_TS_OFFSET_ADJUSTMENT 0 #define DEFAULT_DO_LOST FALSE #define DEFAULT_POST_DROP_MESSAGES FALSE #define DEFAULT_DROP_MESSAGES_INTERVAL_MS 200 #define DEFAULT_MODE RTP_JITTER_BUFFER_MODE_SLAVE #define DEFAULT_PERCENT 0 #define DEFAULT_DO_RETRANSMISSION FALSE #define DEFAULT_RTX_NEXT_SEQNUM TRUE #define DEFAULT_RTX_DELAY -1 #define DEFAULT_RTX_MIN_DELAY 0 #define DEFAULT_RTX_DELAY_REORDER 3 #define DEFAULT_RTX_RETRY_TIMEOUT -1 #define DEFAULT_RTX_MIN_RETRY_TIMEOUT -1 #define DEFAULT_RTX_RETRY_PERIOD -1 #define DEFAULT_RTX_MAX_RETRIES -1 #define DEFAULT_RTX_DEADLINE -1 #define DEFAULT_RTX_STATS_TIMEOUT 1000 #define DEFAULT_MAX_RTCP_RTP_TIME_DIFF 1000 #define DEFAULT_MAX_DROPOUT_TIME 60000 #define DEFAULT_MAX_MISORDER_TIME 2000 #define DEFAULT_RFC7273_SYNC FALSE #define DEFAULT_FASTSTART_MIN_PACKETS 0 #define DEFAULT_AUTO_RTX_DELAY (20 * GST_MSECOND) #define DEFAULT_AUTO_RTX_TIMEOUT (40 * GST_MSECOND) enum { PROP_0, PROP_LATENCY, PROP_DROP_ON_LATENCY, PROP_TS_OFFSET, PROP_MAX_TS_OFFSET_ADJUSTMENT, PROP_DO_LOST, PROP_POST_DROP_MESSAGES, PROP_DROP_MESSAGES_INTERVAL, PROP_MODE, PROP_PERCENT, PROP_DO_RETRANSMISSION, PROP_RTX_NEXT_SEQNUM, PROP_RTX_DELAY, PROP_RTX_MIN_DELAY, PROP_RTX_DELAY_REORDER, PROP_RTX_RETRY_TIMEOUT, PROP_RTX_MIN_RETRY_TIMEOUT, PROP_RTX_RETRY_PERIOD, PROP_RTX_MAX_RETRIES, PROP_RTX_DEADLINE, PROP_RTX_STATS_TIMEOUT, PROP_STATS, PROP_MAX_RTCP_RTP_TIME_DIFF, PROP_MAX_DROPOUT_TIME, PROP_MAX_MISORDER_TIME, PROP_RFC7273_SYNC, PROP_FASTSTART_MIN_PACKETS }; #define JBUF_LOCK(priv) G_STMT_START { \ GST_TRACE("Locking from thread %p", g_thread_self()); \ (g_mutex_lock (&(priv)->jbuf_lock)); \ GST_TRACE("Locked from thread %p", g_thread_self()); \ } G_STMT_END #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_STMT_START { \ GST_TRACE ("Unlocking from thread %p", g_thread_self ()); \ (g_mutex_unlock (&(priv)->jbuf_lock)); \ } G_STMT_END #define JBUF_WAIT_QUEUE(priv) G_STMT_START { \ GST_DEBUG ("waiting queue"); \ (priv)->waiting_queue++; \ g_cond_wait (&(priv)->jbuf_queue, &(priv)->jbuf_lock); \ (priv)->waiting_queue--; \ GST_DEBUG ("waiting queue done"); \ } G_STMT_END #define JBUF_SIGNAL_QUEUE(priv) G_STMT_START { \ if (G_UNLIKELY ((priv)->waiting_queue)) { \ GST_DEBUG ("signal queue, %d waiters", (priv)->waiting_queue); \ g_cond_signal (&(priv)->jbuf_queue); \ } \ } G_STMT_END #define JBUF_WAIT_TIMER(priv) G_STMT_START { \ GST_DEBUG ("waiting timer"); \ (priv)->waiting_timer++; \ g_cond_wait (&(priv)->jbuf_timer, &(priv)->jbuf_lock); \ (priv)->waiting_timer--; \ GST_DEBUG ("waiting timer done"); \ } G_STMT_END #define JBUF_SIGNAL_TIMER(priv) G_STMT_START { \ if (G_UNLIKELY ((priv)->waiting_timer)) { \ GST_DEBUG ("signal timer, %d waiters", (priv)->waiting_timer); \ g_cond_signal (&(priv)->jbuf_timer); \ } \ } G_STMT_END #define JBUF_WAIT_EVENT(priv,label) G_STMT_START { \ GST_DEBUG ("waiting event"); \ (priv)->waiting_event = TRUE; \ g_cond_wait (&(priv)->jbuf_event, &(priv)->jbuf_lock); \ (priv)->waiting_event = FALSE; \ GST_DEBUG ("waiting event done"); \ if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \ goto label; \ } G_STMT_END #define JBUF_SIGNAL_EVENT(priv) G_STMT_START { \ if (G_UNLIKELY ((priv)->waiting_event)) { \ GST_DEBUG ("signal event"); \ g_cond_signal (&(priv)->jbuf_event); \ } \ } G_STMT_END #define JBUF_WAIT_QUERY(priv,label) G_STMT_START { \ GST_DEBUG ("waiting query"); \ (priv)->waiting_query = TRUE; \ g_cond_wait (&(priv)->jbuf_query, &(priv)->jbuf_lock); \ (priv)->waiting_query = FALSE; \ GST_DEBUG ("waiting query done"); \ if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \ goto label; \ } G_STMT_END #define JBUF_SIGNAL_QUERY(priv,res) G_STMT_START { \ (priv)->last_query = res; \ if (G_UNLIKELY ((priv)->waiting_query)) { \ GST_DEBUG ("signal query"); \ g_cond_signal (&(priv)->jbuf_query); \ } \ } G_STMT_END #define GST_BUFFER_IS_RETRANSMISSION(buffer) \ GST_BUFFER_FLAG_IS_SET (buffer, GST_RTP_BUFFER_FLAG_RETRANSMISSION) struct _GstRtpJitterBufferPrivate { GstPad *sinkpad, *srcpad; GstPad *rtcpsinkpad; RTPJitterBuffer *jbuf; GMutex jbuf_lock; guint waiting_queue; GCond jbuf_queue; guint waiting_timer; GCond jbuf_timer; gboolean waiting_event; GCond jbuf_event; gboolean waiting_query; GCond jbuf_query; gboolean last_query; gboolean discont; gboolean ts_discont; gboolean active; guint64 out_offset; guint32 segment_seqnum; gboolean timer_running; GThread *timer_thread; /* properties */ guint latency_ms; guint64 latency_ns; gboolean drop_on_latency; gint64 ts_offset; guint64 max_ts_offset_adjustment; gboolean do_lost; gboolean post_drop_messages; guint drop_messages_interval_ms; gboolean do_retransmission; gboolean rtx_next_seqnum; gint rtx_delay; guint rtx_min_delay; gint rtx_delay_reorder; gint rtx_retry_timeout; gint rtx_min_retry_timeout; gint rtx_retry_period; gint rtx_max_retries; guint rtx_stats_timeout; gint rtx_deadline_ms; gint max_rtcp_rtp_time_diff; guint32 max_dropout_time; guint32 max_misorder_time; guint faststart_min_packets; /* the last seqnum we pushed out */ guint32 last_popped_seqnum; /* the next expected seqnum we push */ guint32 next_seqnum; /* seqnum-base, if known */ guint32 seqnum_base; /* last output time */ GstClockTime last_out_time; /* last valid input timestamp and rtptime pair */ GstClockTime ips_pts; guint64 ips_rtptime; GstClockTime packet_spacing; gint equidistant; GQueue gap_packets; /* the next expected seqnum we receive */ GstClockTime last_in_pts; guint32 next_in_seqnum; /* "normal" timers */ RtpTimerQueue *timers; /* timers used for RTX statistics backlog */ RtpTimerQueue *rtx_stats_timers; /* start and stop ranges */ GstClockTime npt_start; GstClockTime npt_stop; guint64 ext_timestamp; guint64 last_elapsed; guint64 estimated_eos; GstClockID eos_id; /* state */ gboolean eos; guint last_percent; /* clock rate and rtp timestamp offset */ gint last_pt; gint32 clock_rate; gint64 clock_base; gint64 ts_offset_remainder; /* when we are shutting down */ GstFlowReturn srcresult; gboolean blocked; /* for sync */ GstSegment segment; GstClockID clock_id; GstClockTime timer_timeout; guint16 timer_seqnum; /* the latency of the upstream peer, we have to take this into account when * synchronizing the buffers. */ GstClockTime peer_latency; guint64 ext_rtptime; GstBuffer *last_sr; /* some accounting */ guint64 num_pushed; guint64 num_lost; guint64 num_late; guint64 num_duplicates; guint64 num_rtx_requests; guint64 num_rtx_success; guint64 num_rtx_failed; gdouble avg_rtx_num; guint64 avg_rtx_rtt; RTPPacketRateCtx packet_rate_ctx; /* for the jitter */ GstClockTime last_dts; GstClockTime last_pts; guint64 last_rtptime; GstClockTime avg_jitter; /* for dropped packet messages */ GstClockTime last_drop_msg_timestamp; /* accumulators; reset every time a drop message is posted */ guint num_too_late; guint num_drop_on_latency; }; typedef enum { REASON_TOO_LATE, REASON_DROP_ON_LATENCY } DropMessageReason; 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 }; #define gst_rtp_jitter_buffer_parent_class parent_class G_DEFINE_TYPE_WITH_PRIVATE (GstRtpJitterBuffer, gst_rtp_jitter_buffer, 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, const GstCaps * filter); static void gst_rtp_jitter_buffer_release_pad (GstElement * element, GstPad * pad); static GstClock *gst_rtp_jitter_buffer_provide_clock (GstElement * element); static gboolean gst_rtp_jitter_buffer_set_clock (GstElement * element, GstClock * clock); /* pad overrides */ static GstCaps *gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter); static GstIterator *gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad, GstObject * parent); /* sinkpad overrides */ static gboolean gst_rtp_jitter_buffer_sink_event (GstPad * pad, GstObject * parent, GstEvent * event); static GstFlowReturn gst_rtp_jitter_buffer_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer); static GstFlowReturn gst_rtp_jitter_buffer_chain_list (GstPad * pad, GstObject * parent, GstBufferList * buffer_list); static gboolean gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad, GstObject * parent, GstEvent * event); static GstFlowReturn gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad, GstObject * parent, GstBuffer * buffer); static gboolean gst_rtp_jitter_buffer_sink_query (GstPad * pad, GstObject * parent, GstQuery * query); /* srcpad overrides */ static gboolean gst_rtp_jitter_buffer_src_event (GstPad * pad, GstObject * parent, GstEvent * event); static gboolean gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad, GstObject * parent, GstPadMode mode, gboolean active); static void gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer); static gboolean gst_rtp_jitter_buffer_src_query (GstPad * pad, GstObject * parent, 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 do_handle_sync (GstRtpJitterBuffer * jitterbuffer); static void unschedule_current_timer (GstRtpJitterBuffer * jitterbuffer); static void wait_next_timeout (GstRtpJitterBuffer * jitterbuffer); static GstStructure *gst_rtp_jitter_buffer_create_stats (GstRtpJitterBuffer * jitterbuffer); static void update_rtx_stats (GstRtpJitterBuffer * jitterbuffer, const RtpTimer * timer, GstClockTime dts, gboolean success); static GstClockTime get_current_running_time (GstRtpJitterBuffer * jitterbuffer); static void gst_rtp_jitter_buffer_class_init (GstRtpJitterBufferClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; 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:max-ts-offset-adjustment: * * The maximum number of nanoseconds per frame that time offset may be * adjusted with. This is used to avoid sudden large changes to time stamps. */ g_object_class_install_property (gobject_class, PROP_MAX_TS_OFFSET_ADJUSTMENT, g_param_spec_uint64 ("max-ts-offset-adjustment", "Max Timestamp Offset Adjustment", "The maximum number of nanoseconds per frame that time stamp " "offsets may be adjusted (0 = no limit).", 0, G_MAXUINT64, DEFAULT_MAX_TS_OFFSET_ADJUSTMENT, 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:post-drop-messages: * * Post custom messages to the bus when a packet is dropped by the * jitterbuffer due to arriving too late, being already considered lost, * or being dropped due to the drop-on-latency property being enabled. * Message is of type GST_MESSAGE_ELEMENT and contains a GstStructure named * "drop-msg" with the following fields: * * * #guint `seqnum`: Seqnum of dropped packet. * * #guint64 `timestamp`: PTS timestamp of dropped packet. * * #gstring `reason`: Reason for dropping the packet. * * #guint `num-too-late`: Number of packets arriving too late since * last drop message. * * #guint `num-drop-on-latency`: Number of packets dropped due to the * drop-on-latency property since last drop message. * * Since: 1.18 */ g_object_class_install_property (gobject_class, PROP_POST_DROP_MESSAGES, g_param_spec_boolean ("post-drop-messages", "Post drop messages", "Post a custom message to the bus when a packet is dropped by the jitterbuffer", DEFAULT_POST_DROP_MESSAGES, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRtpJitterBuffer:drop-messages-interval: * * Minimal time in milliseconds between posting dropped packet messages, if enabled * by setting property by setting #GstRtpJitterBuffer:post-drop-messages to %TRUE. * If interval is set to 0, every dropped packet will result in a drop message being posted. * * Since: 1.18 */ g_object_class_install_property (gobject_class, PROP_DROP_MESSAGES_INTERVAL, g_param_spec_uint ("drop-messages-interval", "Drop message interval", "Minimal time between posting dropped packet messages", 0, G_MAXUINT, DEFAULT_DROP_MESSAGES_INTERVAL_MS, 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. */ 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:do-retransmission: * * Send out a GstRTPRetransmission event upstream when a packet is considered * late and should be retransmitted. * * Since: 1.2 */ g_object_class_install_property (gobject_class, PROP_DO_RETRANSMISSION, g_param_spec_boolean ("do-retransmission", "Do Retransmission", "Send retransmission events upstream when a packet is late", DEFAULT_DO_RETRANSMISSION, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRtpJitterBuffer:rtx-next-seqnum * * Estimate when the next packet should arrive and schedule a retransmission * request for it. * This is, when packet N arrives, a GstRTPRetransmission event is schedule * for packet N+1. So it will be requested if it does not arrive at the expected time. * The expected time is calculated using the dts of N and the packet spacing. * * Since: 1.6 */ g_object_class_install_property (gobject_class, PROP_RTX_NEXT_SEQNUM, g_param_spec_boolean ("rtx-next-seqnum", "RTX next seqnum", "Estimate when the next packet should arrive and schedule a " "retransmission request for it.", DEFAULT_RTX_NEXT_SEQNUM, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRtpJitterBuffer:rtx-delay: * * When a packet did not arrive at the expected time, wait this extra amount * of time before sending a retransmission event. * * When -1 is used, the max jitter will be used as extra delay. * * Since: 1.2 */ g_object_class_install_property (gobject_class, PROP_RTX_DELAY, g_param_spec_int ("rtx-delay", "RTX Delay", "Extra time in ms to wait before sending retransmission " "event (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_DELAY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRtpJitterBuffer:rtx-min-delay: * * When a packet did not arrive at the expected time, wait at least this extra amount * of time before sending a retransmission event. * * Since: 1.6 */ g_object_class_install_property (gobject_class, PROP_RTX_MIN_DELAY, g_param_spec_uint ("rtx-min-delay", "Minimum RTX Delay", "Minimum time in ms to wait before sending retransmission " "event", 0, G_MAXUINT, DEFAULT_RTX_MIN_DELAY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRtpJitterBuffer:rtx-delay-reorder: * * Assume that a retransmission event should be sent when we see * this much packet reordering. * * When -1 is used, the value will be estimated based on observed packet * reordering. When 0 is used packet reordering alone will not cause a * retransmission event (Since 1.10). * * Since: 1.2 */ g_object_class_install_property (gobject_class, PROP_RTX_DELAY_REORDER, g_param_spec_int ("rtx-delay-reorder", "RTX Delay Reorder", "Sending retransmission event when this much reordering " "(0 disable)", -1, G_MAXINT, DEFAULT_RTX_DELAY_REORDER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRtpJitterBuffer::rtx-retry-timeout: * * When no packet has been received after sending a retransmission event * for this time, retry sending a retransmission event. * * When -1 is used, the value will be estimated based on observed round * trip time. * * Since: 1.2 */ g_object_class_install_property (gobject_class, PROP_RTX_RETRY_TIMEOUT, g_param_spec_int ("rtx-retry-timeout", "RTX Retry Timeout", "Retry sending a transmission event after this timeout in " "ms (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_RETRY_TIMEOUT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRtpJitterBuffer::rtx-min-retry-timeout: * * The minimum amount of time between retry timeouts. When * GstRtpJitterBuffer::rtx-retry-timeout is -1, this value ensures a * minimum interval between retry timeouts. * * When -1 is used, the value will be estimated based on the * packet spacing. * * Since: 1.6 */ g_object_class_install_property (gobject_class, PROP_RTX_MIN_RETRY_TIMEOUT, g_param_spec_int ("rtx-min-retry-timeout", "RTX Min Retry Timeout", "Minimum timeout between sending a transmission event in " "ms (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_MIN_RETRY_TIMEOUT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRtpJitterBuffer:rtx-retry-period: * * The amount of time to try to get a retransmission. * * When -1 is used, the value will be estimated based on the jitterbuffer * latency and the observed round trip time. * * Since: 1.2 */ g_object_class_install_property (gobject_class, PROP_RTX_RETRY_PERIOD, g_param_spec_int ("rtx-retry-period", "RTX Retry Period", "Try to get a retransmission for this many ms " "(-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_RETRY_PERIOD, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRtpJitterBuffer:rtx-max-retries: * * The maximum number of retries to request a retransmission. * * This implies that as maximum (rtx-max-retries + 1) retransmissions will be requested. * When -1 is used, the number of retransmission request will not be limited. * * Since: 1.6 */ g_object_class_install_property (gobject_class, PROP_RTX_MAX_RETRIES, g_param_spec_int ("rtx-max-retries", "RTX Max Retries", "The maximum number of retries to request a retransmission. " "(-1 not limited)", -1, G_MAXINT, DEFAULT_RTX_MAX_RETRIES, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRtpJitterBuffer:rtx-deadline: * * The deadline for a valid RTX request in ms. * * How long the RTX RTCP will be valid for. * When -1 is used, the size of the jitterbuffer will be used. * * Since: 1.10 */ g_object_class_install_property (gobject_class, PROP_RTX_DEADLINE, g_param_spec_int ("rtx-deadline", "RTX Deadline (ms)", "The deadline for a valid RTX request in milliseconds. " "(-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_DEADLINE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRtpJitterBuffer::rtx-stats-timeout: * * The time to wait for a retransmitted packet after it has been * considered lost in order to collect RTX statistics. * * Since: 1.10 */ g_object_class_install_property (gobject_class, PROP_RTX_STATS_TIMEOUT, g_param_spec_uint ("rtx-stats-timeout", "RTX Statistics Timeout", "The time to wait for a retransmitted packet after it has been " "considered lost in order to collect statistics (ms)", 0, G_MAXUINT, DEFAULT_RTX_STATS_TIMEOUT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_MAX_DROPOUT_TIME, g_param_spec_uint ("max-dropout-time", "Max dropout time", "The maximum time (milliseconds) of missing packets tolerated.", 0, G_MAXINT32, DEFAULT_MAX_DROPOUT_TIME, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_MAX_MISORDER_TIME, g_param_spec_uint ("max-misorder-time", "Max misorder time", "The maximum time (milliseconds) of misordered packets tolerated.", 0, G_MAXUINT, DEFAULT_MAX_MISORDER_TIME, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRtpJitterBuffer:stats: * * Various jitterbuffer statistics. This property returns a GstStructure * with name application/x-rtp-jitterbuffer-stats with the following fields: * * * #guint64 `num-pushed`: the number of packets pushed out. * * #guint64 `num-lost`: the number of packets considered lost. * * #guint64 `num-late`: the number of packets arriving too late. * * #guint64 `num-duplicates`: the number of duplicate packets. * * #guint64 `rtx-count`: the number of retransmissions requested. * * #guint64 `rtx-success-count`: the number of successful retransmissions. * * #gdouble `rtx-per-packet`: average number of RTX per packet. * * #guint64 `rtx-rtt`: average round trip time per RTX. * * Since: 1.4 */ g_object_class_install_property (gobject_class, PROP_STATS, g_param_spec_boxed ("stats", "Statistics", "Various statistics", GST_TYPE_STRUCTURE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); /** * GstRtpJitterBuffer:max-rtcp-rtp-time-diff * * The maximum amount of time in ms that the RTP time in the RTCP SRs * is allowed to be ahead of the last RTP packet we received. Use * -1 to disable ignoring of RTCP packets. * * Since: 1.8 */ g_object_class_install_property (gobject_class, PROP_MAX_RTCP_RTP_TIME_DIFF, g_param_spec_int ("max-rtcp-rtp-time-diff", "Max RTCP RTP Time Diff", "Maximum amount of time in ms that the RTP time in RTCP SRs " "is allowed to be ahead (-1 disabled)", -1, G_MAXINT, DEFAULT_MAX_RTCP_RTP_TIME_DIFF, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_RFC7273_SYNC, g_param_spec_boolean ("rfc7273-sync", "Sync on RFC7273 clock", "Synchronize received streams to the RFC7273 clock " "(requires clock and offset to be provided)", DEFAULT_RFC7273_SYNC, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRtpJitterBuffer:faststart-min-packets * * The number of consecutive packets needed to start (set to 0 to * disable faststart. The jitterbuffer will by default start after the * latency has elapsed) * * Since: 1.14 */ g_object_class_install_property (gobject_class, PROP_FASTSTART_MIN_PACKETS, g_param_spec_uint ("faststart-min-packets", "Faststart minimum packets", "The number of consecutive packets needed to start (set to 0 to " "disable faststart. The jitterbuffer will by default start after " "the latency has elapsed)", 0, G_MAXUINT, DEFAULT_FASTSTART_MIN_PACKETS, G_PARAM_READWRITE | 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, NULL, 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, NULL, 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, NULL, 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, NULL, 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. */ 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, NULL, 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); gstelement_class->set_clock = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_set_clock); gst_element_class_add_static_pad_template (gstelement_class, &gst_rtp_jitter_buffer_src_template); gst_element_class_add_static_pad_template (gstelement_class, &gst_rtp_jitter_buffer_sink_template); gst_element_class_add_static_pad_template (gstelement_class, &gst_rtp_jitter_buffer_sink_rtcp_template); gst_element_class_set_static_metadata (gstelement_class, "RTP packet jitter-buffer", "Filter/Network/RTP", "A buffer that deals with network jitter and other transmission faults", "Philippe Kalaf , " "Wim Taymans "); 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, "rtpjitterbuffer", 0, "RTP Jitter Buffer"); GST_DEBUG_REGISTER_FUNCPTR (gst_rtp_jitter_buffer_chain_rtcp); } static void gst_rtp_jitter_buffer_init (GstRtpJitterBuffer * jitterbuffer) { GstRtpJitterBufferPrivate *priv; priv = gst_rtp_jitter_buffer_get_instance_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->ts_offset = DEFAULT_TS_OFFSET; priv->max_ts_offset_adjustment = DEFAULT_MAX_TS_OFFSET_ADJUSTMENT; priv->do_lost = DEFAULT_DO_LOST; priv->post_drop_messages = DEFAULT_POST_DROP_MESSAGES; priv->drop_messages_interval_ms = DEFAULT_DROP_MESSAGES_INTERVAL_MS; priv->do_retransmission = DEFAULT_DO_RETRANSMISSION; priv->rtx_next_seqnum = DEFAULT_RTX_NEXT_SEQNUM; priv->rtx_delay = DEFAULT_RTX_DELAY; priv->rtx_min_delay = DEFAULT_RTX_MIN_DELAY; priv->rtx_delay_reorder = DEFAULT_RTX_DELAY_REORDER; priv->rtx_retry_timeout = DEFAULT_RTX_RETRY_TIMEOUT; priv->rtx_min_retry_timeout = DEFAULT_RTX_MIN_RETRY_TIMEOUT; priv->rtx_retry_period = DEFAULT_RTX_RETRY_PERIOD; priv->rtx_max_retries = DEFAULT_RTX_MAX_RETRIES; priv->rtx_deadline_ms = DEFAULT_RTX_DEADLINE; priv->rtx_stats_timeout = DEFAULT_RTX_STATS_TIMEOUT; priv->max_rtcp_rtp_time_diff = DEFAULT_MAX_RTCP_RTP_TIME_DIFF; priv->max_dropout_time = DEFAULT_MAX_DROPOUT_TIME; priv->max_misorder_time = DEFAULT_MAX_MISORDER_TIME; priv->faststart_min_packets = DEFAULT_FASTSTART_MIN_PACKETS; priv->ts_offset_remainder = 0; priv->last_dts = -1; priv->last_pts = -1; priv->last_rtptime = -1; priv->avg_jitter = 0; priv->last_drop_msg_timestamp = GST_CLOCK_TIME_NONE; priv->num_too_late = 0; priv->num_drop_on_latency = 0; priv->segment_seqnum = GST_SEQNUM_INVALID; priv->timers = rtp_timer_queue_new (); priv->rtx_stats_timers = rtp_timer_queue_new (); priv->jbuf = rtp_jitter_buffer_new (); g_mutex_init (&priv->jbuf_lock); g_cond_init (&priv->jbuf_queue); g_cond_init (&priv->jbuf_timer); g_cond_init (&priv->jbuf_event); g_cond_init (&priv->jbuf_query); g_queue_init (&priv->gap_packets); gst_segment_init (&priv->segment, GST_FORMAT_TIME); /* reset skew detection initially */ 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_activatemode_function (priv->srcpad, GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_activate_mode)); gst_pad_set_query_function (priv->srcpad, GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_query)); 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_chain_list_function (priv->sinkpad, GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_chain_list)); gst_pad_set_event_function (priv->sinkpad, GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_event)); gst_pad_set_query_function (priv->sinkpad, GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_query)); gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->srcpad); gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->sinkpad); GST_OBJECT_FLAG_SET (jitterbuffer, GST_ELEMENT_FLAG_PROVIDE_CLOCK); } static void free_item_and_retain_sticky_events (RTPJitterBufferItem * item, gpointer user_data) { GList **l = user_data; if (item->data && item->type == ITEM_TYPE_EVENT && GST_EVENT_IS_STICKY (item->data)) { *l = g_list_prepend (*l, item->data); item->data = NULL; } rtp_jitter_buffer_free_item (item); } static void gst_rtp_jitter_buffer_finalize (GObject * object) { GstRtpJitterBuffer *jitterbuffer; GstRtpJitterBufferPrivate *priv; jitterbuffer = GST_RTP_JITTER_BUFFER (object); priv = jitterbuffer->priv; g_object_unref (priv->timers); g_object_unref (priv->rtx_stats_timers); g_mutex_clear (&priv->jbuf_lock); g_cond_clear (&priv->jbuf_queue); g_cond_clear (&priv->jbuf_timer); g_cond_clear (&priv->jbuf_event); g_cond_clear (&priv->jbuf_query); rtp_jitter_buffer_flush (priv->jbuf, NULL, NULL); g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL); g_queue_clear (&priv->gap_packets); g_object_unref (priv->jbuf); G_OBJECT_CLASS (parent_class)->finalize (object); } static GstIterator * gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad, GstObject * parent) { GstRtpJitterBuffer *jitterbuffer; GstPad *otherpad = NULL; GstIterator *it = NULL; GValue val = { 0, }; jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent); 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) { it = gst_iterator_new_single (GST_TYPE_PAD, NULL); } if (it == NULL) { g_value_init (&val, GST_TYPE_PAD); g_value_set_object (&val, otherpad); it = gst_iterator_new_single (GST_TYPE_PAD, &val); g_value_unset (&val); } 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, const GstCaps * filter) { 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_CAST (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 ("rtpjitterbuffer: this is not our template"); return NULL; } exists: { g_warning ("rtpjitterbuffer: 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_CAST (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 gboolean gst_rtp_jitter_buffer_set_clock (GstElement * element, GstClock * clock) { GstRtpJitterBuffer *jitterbuffer = GST_RTP_JITTER_BUFFER (element); rtp_jitter_buffer_set_pipeline_clock (jitterbuffer->priv->jbuf, clock); return GST_ELEMENT_CLASS (parent_class)->set_clock (element, clock); } 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; /* do not clear current content, but refresh state for new arrival */ GST_DEBUG_OBJECT (jitterbuffer, "reset jitterbuffer"); rtp_jitter_buffer_reset_skew (priv->jbuf); JBUF_UNLOCK (priv); } static GstClockTime gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jbuf, gboolean active, guint64 offset) { GstRtpJitterBufferPrivate *priv; GstClockTime last_out; RTPJitterBufferItem *item; 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_EVENT (priv); } if (!active) { rtp_jitter_buffer_set_buffering (priv->jbuf, TRUE); } if ((item = rtp_jitter_buffer_peek (priv->jbuf))) { /* head buffer timestamp and offset gives our output time */ last_out = item->pts + 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, GstCaps * filter) { GstRtpJitterBuffer *jitterbuffer; GstRtpJitterBufferPrivate *priv; GstPad *other; GstCaps *caps; 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_query_caps (other, filter); templ = gst_pad_get_pad_template_caps (pad); if (caps == NULL) { GST_DEBUG_OBJECT (jitterbuffer, "use template"); caps = templ; } else { GstCaps *intersect; GST_DEBUG_OBJECT (jitterbuffer, "intersect with template"); intersect = gst_caps_intersect (caps, templ); gst_caps_unref (caps); gst_caps_unref (templ); caps = intersect; } gst_object_unref (jitterbuffer); return caps; } /* g_ascii_string_to_unsigned is available since 2.54. Get rid of this wrapper * when we bump the version in 1.18 */ #if !GLIB_CHECK_VERSION(2,54,0) #define g_ascii_string_to_unsigned _gst_jitter_buffer_ascii_string_to_unsigned static gboolean _gst_jitter_buffer_ascii_string_to_unsigned (const gchar * str, guint base, guint64 min, guint64 max, guint64 * out_num, GError ** error) { gchar *endptr = NULL; *out_num = g_ascii_strtoull (str, &endptr, base); if (errno) return FALSE; if (endptr == str) return FALSE; return TRUE; } #endif /* * Must be called with JBUF_LOCK held */ static gboolean gst_jitter_buffer_sink_parse_caps (GstRtpJitterBuffer * jitterbuffer, GstCaps * caps, gint pt) { GstRtpJitterBufferPrivate *priv; GstStructure *caps_struct; guint val; gint payload = -1; GstClockTime tval; const gchar *ts_refclk, *mediaclk; priv = jitterbuffer->priv; /* first parse the caps */ caps_struct = gst_caps_get_structure (caps, 0); GST_DEBUG_OBJECT (jitterbuffer, "got caps %" GST_PTR_FORMAT, caps); if (gst_structure_get_int (caps_struct, "payload", &payload) && pt != -1 && payload != pt) { GST_ERROR_OBJECT (jitterbuffer, "Got caps with wrong payload type (got %d, expected %d)", pt, payload); return FALSE; } if (payload != -1) { GST_DEBUG_OBJECT (jitterbuffer, "Got payload type %d", payload); priv->last_pt = payload; } /* 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); rtp_jitter_buffer_set_clock_rate (priv->jbuf, priv->clock_rate); gst_rtp_packet_rate_ctx_reset (&priv->packet_rate_ctx, 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; JBUF_SIGNAL_EVENT (priv); } priv->seqnum_base = val; } else { priv->seqnum_base = -1; } 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)); if ((ts_refclk = gst_structure_get_string (caps_struct, "a-ts-refclk"))) { GstClock *clock = NULL; guint64 clock_offset = -1; GST_DEBUG_OBJECT (jitterbuffer, "Have timestamp reference clock %s", ts_refclk); if (g_str_has_prefix (ts_refclk, "ntp=")) { if (g_str_has_prefix (ts_refclk, "ntp=/traceable/")) { GST_FIXME_OBJECT (jitterbuffer, "Can't handle traceable NTP clocks"); } else { const gchar *host, *portstr; gchar *hostname; guint port; host = ts_refclk + sizeof ("ntp=") - 1; if (host[0] == '[') { /* IPv6 */ portstr = strchr (host, ']'); if (portstr && portstr[1] == ':') portstr = portstr + 1; else portstr = NULL; } else { portstr = strrchr (host, ':'); } if (!portstr || sscanf (portstr, ":%u", &port) != 1) port = 123; if (portstr) hostname = g_strndup (host, (portstr - host)); else hostname = g_strdup (host); clock = gst_ntp_clock_new (NULL, hostname, port, 0); g_free (hostname); } } else if (g_str_has_prefix (ts_refclk, "ptp=IEEE1588-2008:")) { const gchar *domainstr = ts_refclk + sizeof ("ptp=IEEE1588-2008:XX-XX-XX-XX-XX-XX-XX-XX") - 1; guint domain; if (domainstr[0] != ':' || sscanf (domainstr, ":%u", &domain) != 1) domain = 0; clock = gst_ptp_clock_new (NULL, domain); } else { GST_FIXME_OBJECT (jitterbuffer, "Unsupported timestamp reference clock"); } if ((mediaclk = gst_structure_get_string (caps_struct, "a-mediaclk"))) { GST_DEBUG_OBJECT (jitterbuffer, "Got media clock %s", mediaclk); if (!g_str_has_prefix (mediaclk, "direct=") || !g_ascii_string_to_unsigned (&mediaclk[8], 10, 0, G_MAXUINT64, &clock_offset, NULL)) GST_FIXME_OBJECT (jitterbuffer, "Unsupported media clock"); if (strstr (mediaclk, "rate=") != NULL) { GST_FIXME_OBJECT (jitterbuffer, "Rate property not supported"); clock_offset = -1; } } rtp_jitter_buffer_set_media_clock (priv->jbuf, clock, clock_offset); } else { rtp_jitter_buffer_set_media_clock (priv->jbuf, NULL, -1); } 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 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_FLUSHING; GST_DEBUG_OBJECT (jitterbuffer, "Disabling pop on queue"); /* this unblocks any waiting pops on the src pad task */ JBUF_SIGNAL_EVENT (priv); JBUF_SIGNAL_QUERY (priv, FALSE); JBUF_SIGNAL_QUEUE (priv); 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 = GST_CLOCK_TIME_NONE; priv->next_seqnum = -1; priv->seqnum_base = -1; priv->ips_rtptime = -1; priv->ips_pts = GST_CLOCK_TIME_NONE; priv->packet_spacing = 0; priv->next_in_seqnum = -1; priv->clock_rate = -1; priv->last_pt = -1; priv->eos = FALSE; priv->estimated_eos = -1; priv->last_elapsed = 0; priv->ext_timestamp = -1; priv->avg_jitter = 0; priv->last_dts = -1; priv->last_rtptime = -1; priv->last_in_pts = 0; priv->equidistant = 0; priv->segment_seqnum = GST_SEQNUM_INVALID; priv->last_drop_msg_timestamp = GST_CLOCK_TIME_NONE; priv->num_too_late = 0; priv->num_drop_on_latency = 0; GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer"); rtp_jitter_buffer_flush (priv->jbuf, NULL, NULL); rtp_jitter_buffer_disable_buffering (priv->jbuf, FALSE); rtp_jitter_buffer_reset_skew (priv->jbuf); rtp_timer_queue_remove_all (priv->timers); g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL); g_queue_clear (&priv->gap_packets); JBUF_UNLOCK (priv); } static gboolean gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad, GstObject * parent, GstPadMode mode, gboolean active) { gboolean result; GstRtpJitterBuffer *jitterbuffer = NULL; jitterbuffer = GST_RTP_JITTER_BUFFER (parent); switch (mode) { case GST_PAD_MODE_PUSH: if (active) { /* allow data processing */ gst_rtp_jitter_buffer_flush_stop (jitterbuffer); /* start pushing out buffers */ GST_DEBUG_OBJECT (jitterbuffer, "Starting task on srcpad"); result = gst_pad_start_task (jitterbuffer->priv->srcpad, (GstTaskFunction) gst_rtp_jitter_buffer_loop, jitterbuffer, NULL); } 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); } break; default: result = FALSE; break; } 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; priv->timer_running = TRUE; priv->srcresult = GST_FLOW_OK; priv->timer_thread = g_thread_new ("timer", (GThreadFunc) wait_next_timeout, jitterbuffer); 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_EVENT (priv); JBUF_SIGNAL_TIMER (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; unschedule_current_timer (jitterbuffer); JBUF_UNLOCK (priv); if (ret != GST_STATE_CHANGE_FAILURE) ret = GST_STATE_CHANGE_NO_PREROLL; break; case GST_STATE_CHANGE_PAUSED_TO_READY: JBUF_LOCK (priv); gst_buffer_replace (&priv->last_sr, NULL); priv->timer_running = FALSE; priv->srcresult = GST_FLOW_FLUSHING; unschedule_current_timer (jitterbuffer); JBUF_SIGNAL_TIMER (priv); JBUF_SIGNAL_QUERY (priv, FALSE); JBUF_SIGNAL_QUEUE (priv); JBUF_UNLOCK (priv); g_thread_join (priv->timer_thread); priv->timer_thread = NULL; break; case GST_STATE_CHANGE_READY_TO_NULL: break; default: break; } return ret; } static gboolean gst_rtp_jitter_buffer_src_event (GstPad * pad, GstObject * parent, GstEvent * event) { gboolean ret = TRUE; GstRtpJitterBuffer *jitterbuffer; GstRtpJitterBufferPrivate *priv; jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent); 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); GST_DEBUG_OBJECT (jitterbuffer, "configuring latency of %" GST_TIME_FORMAT, GST_TIME_ARGS (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; } return ret; } /* handles and stores the event in the jitterbuffer, must be called with * LOCK */ static gboolean queue_event (GstRtpJitterBuffer * jitterbuffer, GstEvent * event) { GstRtpJitterBufferPrivate *priv = jitterbuffer->priv; gboolean head; switch (GST_EVENT_TYPE (event)) { case GST_EVENT_CAPS: { GstCaps *caps; gst_event_parse_caps (event, &caps); gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps, -1); break; } case GST_EVENT_SEGMENT: { GstSegment segment; gst_event_copy_segment (event, &segment); priv->segment_seqnum = gst_event_get_seqnum (event); /* we need time for now */ if (segment.format != GST_FORMAT_TIME) { GST_DEBUG_OBJECT (jitterbuffer, "ignoring non-TIME newsegment"); gst_event_unref (event); gst_segment_init (&segment, GST_FORMAT_TIME); event = gst_event_new_segment (&segment); gst_event_set_seqnum (event, priv->segment_seqnum); } priv->segment = segment; break; } case GST_EVENT_EOS: priv->eos = TRUE; rtp_jitter_buffer_disable_buffering (priv->jbuf, TRUE); break; default: break; } GST_DEBUG_OBJECT (jitterbuffer, "adding event"); head = rtp_jitter_buffer_append_event (priv->jbuf, event); if (head || priv->eos) JBUF_SIGNAL_EVENT (priv); return TRUE; } static gboolean gst_rtp_jitter_buffer_sink_event (GstPad * pad, GstObject * parent, GstEvent * event) { gboolean ret = TRUE; GstRtpJitterBuffer *jitterbuffer; GstRtpJitterBufferPrivate *priv; jitterbuffer = GST_RTP_JITTER_BUFFER (parent); priv = jitterbuffer->priv; GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event)); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_FLUSH_START: ret = gst_pad_push_event (priv->srcpad, event); gst_rtp_jitter_buffer_flush_start (jitterbuffer); /* wait for the loop to go into PAUSED */ gst_pad_pause_task (priv->srcpad); break; case GST_EVENT_FLUSH_STOP: ret = gst_pad_push_event (priv->srcpad, event); ret = gst_rtp_jitter_buffer_src_activate_mode (priv->srcpad, parent, GST_PAD_MODE_PUSH, TRUE); break; default: if (GST_EVENT_IS_SERIALIZED (event)) { /* serialized events go in the queue */ JBUF_LOCK (priv); if (priv->srcresult != GST_FLOW_OK) { /* Errors in sticky event pushing are no problem and ignored here * as they will cause more meaningful errors during data flow. * For EOS events, that are not followed by data flow, we still * return FALSE here though. */ if (!GST_EVENT_IS_STICKY (event) || GST_EVENT_TYPE (event) == GST_EVENT_EOS) goto out_flow_error; } /* refuse more events on EOS */ if (priv->eos) goto out_eos; ret = queue_event (jitterbuffer, event); JBUF_UNLOCK (priv); } else { /* non-serialized events are forwarded downstream immediately */ ret = gst_pad_push_event (priv->srcpad, event); } break; } return ret; /* ERRORS */ out_flow_error: { GST_DEBUG_OBJECT (jitterbuffer, "refusing event, we have a downstream flow error: %s", gst_flow_get_name (priv->srcresult)); JBUF_UNLOCK (priv); gst_event_unref (event); return FALSE; } out_eos: { GST_DEBUG_OBJECT (jitterbuffer, "refusing event, we are EOS"); JBUF_UNLOCK (priv); gst_event_unref (event); return FALSE; } } static gboolean gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad, GstObject * parent, GstEvent * event) { gboolean ret = TRUE; GstRtpJitterBuffer *jitterbuffer; jitterbuffer = GST_RTP_JITTER_BUFFER (parent); GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event)); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_FLUSH_START: gst_event_unref (event); break; case GST_EVENT_FLUSH_STOP: gst_event_unref (event); break; default: ret = gst_pad_event_default (pad, parent, event); break; } return ret; } /* * Must be called with JBUF_LOCK held, will release the LOCK when emitting the * signal. The function returns GST_FLOW_ERROR when a parsing error happened and * GST_FLOW_FLUSHING 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, pt); 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_FLUSHING; } parse_failed: { GST_DEBUG_OBJECT (jitterbuffer, "parse failed"); return GST_FLOW_ERROR; } } /* call with jbuf lock held */ static GstMessage * check_buffering_percent (GstRtpJitterBuffer * jitterbuffer, gint percent) { GstRtpJitterBufferPrivate *priv = jitterbuffer->priv; GstMessage *message = NULL; if (percent == -1) return NULL; /* Post a buffering message */ if (priv->last_percent != percent) { priv->last_percent = percent; message = gst_message_new_buffering (GST_OBJECT_CAST (jitterbuffer), percent); gst_message_set_buffering_stats (message, GST_BUFFERING_LIVE, -1, -1, -1); } return message; } /* call with jbuf lock held */ static GstMessage * new_drop_message (GstRtpJitterBuffer * jitterbuffer, guint seqnum, GstClockTime timestamp, DropMessageReason reason) { GstRtpJitterBufferPrivate *priv = jitterbuffer->priv; GstMessage *drop_msg = NULL; GstStructure *s; GstClockTime current_time; GstClockTime time_diff; const gchar *reason_str; current_time = get_current_running_time (jitterbuffer); time_diff = current_time - priv->last_drop_msg_timestamp; if (reason == REASON_TOO_LATE) { priv->num_too_late++; reason_str = "too-late"; } else if (reason == REASON_DROP_ON_LATENCY) { priv->num_drop_on_latency++; reason_str = "drop-on-latency"; } else { GST_WARNING_OBJECT (jitterbuffer, "Invalid reason for drop message"); return drop_msg; } /* Only create new drop_msg if time since last drop_msg is larger that * that the set interval, or if it is the first drop message posted */ if ((time_diff >= priv->drop_messages_interval_ms * GST_MSECOND) || (priv->last_drop_msg_timestamp == GST_CLOCK_TIME_NONE)) { s = gst_structure_new ("drop-msg", "seqnum", G_TYPE_UINT, seqnum, "timestamp", GST_TYPE_CLOCK_TIME, timestamp, "reason", G_TYPE_STRING, reason_str, "num-too-late", G_TYPE_UINT, priv->num_too_late, "num-drop-on-latency", G_TYPE_UINT, priv->num_drop_on_latency, NULL); priv->last_drop_msg_timestamp = current_time; priv->num_too_late = 0; priv->num_drop_on_latency = 0; drop_msg = gst_message_new_element (GST_OBJECT (jitterbuffer), s); } return drop_msg; } static inline GstClockTimeDiff timeout_offset (GstRtpJitterBuffer * jitterbuffer) { GstRtpJitterBufferPrivate *priv = jitterbuffer->priv; return priv->ts_offset + priv->out_offset + priv->latency_ns; } static inline GstClockTime get_pts_timeout (const RtpTimer * timer) { if (timer->timeout == -1) return -1; return timer->timeout - timer->offset; } static void update_timer_offsets (GstRtpJitterBuffer * jitterbuffer) { GstRtpJitterBufferPrivate *priv = jitterbuffer->priv; RtpTimer *test = rtp_timer_queue_peek_earliest (priv->timers); GstClockTimeDiff new_offset = timeout_offset (jitterbuffer); while (test) { if (test->type != RTP_TIMER_EXPECTED) { test->timeout = get_pts_timeout (test) + new_offset; test->offset = new_offset; /* as we apply the offset on all timers, the order of timers won't * change and we can skip updating the timer queue */ } test = rtp_timer_get_next (test); } } static void update_offset (GstRtpJitterBuffer * jitterbuffer) { GstRtpJitterBufferPrivate *priv; priv = jitterbuffer->priv; if (priv->ts_offset_remainder != 0) { GST_DEBUG ("adjustment %" G_GUINT64_FORMAT " remain %" G_GINT64_FORMAT " off %" G_GINT64_FORMAT, priv->max_ts_offset_adjustment, priv->ts_offset_remainder, priv->ts_offset); if (ABS (priv->ts_offset_remainder) > priv->max_ts_offset_adjustment) { if (priv->ts_offset_remainder > 0) { priv->ts_offset += priv->max_ts_offset_adjustment; priv->ts_offset_remainder -= priv->max_ts_offset_adjustment; } else { priv->ts_offset -= priv->max_ts_offset_adjustment; priv->ts_offset_remainder += priv->max_ts_offset_adjustment; } } else { priv->ts_offset += priv->ts_offset_remainder; priv->ts_offset_remainder = 0; } update_timer_offsets (jitterbuffer); } } 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 void unschedule_current_timer (GstRtpJitterBuffer * jitterbuffer) { GstRtpJitterBufferPrivate *priv = jitterbuffer->priv; if (priv->clock_id) { GST_DEBUG_OBJECT (jitterbuffer, "unschedule current timer"); gst_clock_id_unschedule (priv->clock_id); priv->clock_id = NULL; } } static void update_current_timer (GstRtpJitterBuffer * jitterbuffer) { GstRtpJitterBufferPrivate *priv = jitterbuffer->priv; RtpTimer *timer; timer = rtp_timer_queue_peek_earliest (priv->timers); /* we never need to wakeup the timer thread when there is no more timers, if * it was waiting on a clock id, it will simply do later and then wait on * the conditions */ if (timer == NULL) { GST_DEBUG_OBJECT (jitterbuffer, "no more timers"); return; } GST_DEBUG_OBJECT (jitterbuffer, "waiting till %" GST_TIME_FORMAT " and earliest timeout is at %" GST_TIME_FORMAT, GST_TIME_ARGS (priv->timer_timeout), GST_TIME_ARGS (timer->timeout)); /* wakeup the timer thread in case the timer queue was empty */ JBUF_SIGNAL_TIMER (priv); /* no need to wait if the current wait is earlier or later */ if (timer->timeout != -1 && timer->timeout >= priv->timer_timeout) return; /* for other cases, force a reschedule of the timer thread */ unschedule_current_timer (jitterbuffer); } /* get the extra delay to wait before sending RTX */ static GstClockTime get_rtx_delay (GstRtpJitterBufferPrivate * priv) { GstClockTime delay; if (priv->rtx_delay == -1) { /* the maximum delay for any RTX-packet is given by the latency, since anything after that is considered lost. For various calulcations, (given large avg_jitter and/or packet_spacing), the resulting delay could exceed the configured latency, ending up issuing an RTX-request that would never arrive in time. To help this we cap the delay for any RTX with the last possible time it could still arrive in time. */ GstClockTime delay_max = (priv->latency_ns > priv->avg_rtx_rtt) ? priv->latency_ns - priv->avg_rtx_rtt : priv->latency_ns; if (priv->avg_jitter == 0 && priv->packet_spacing == 0) { delay = DEFAULT_AUTO_RTX_DELAY; } else { /* jitter is in nanoseconds, maximum of 2x jitter and half the * packet spacing is a good margin */ delay = MAX (priv->avg_jitter * 2, priv->packet_spacing / 2); } delay = MIN (delay_max, delay); } else { delay = priv->rtx_delay * GST_MSECOND; } if (priv->rtx_min_delay > 0) delay = MAX (delay, priv->rtx_min_delay * GST_MSECOND); return delay; } /* we just received a packet with seqnum and dts. * * First check for old seqnum that we are still expecting. If the gap with the * current seqnum is too big, unschedule the timeouts. * * If we have a valid packet spacing estimate we can set a timer for when we * should receive the next packet. * If we don't have a valid estimate, we remove any timer we might have * had for this packet. */ static void update_timers (GstRtpJitterBuffer * jitterbuffer, guint16 seqnum, GstClockTime dts, GstClockTime pts, gboolean do_next_seqnum, gboolean is_rtx, RtpTimer * timer) { GstRtpJitterBufferPrivate *priv = jitterbuffer->priv; gboolean is_stats_timer = FALSE; if (timer && rtp_timer_queue_find (priv->rtx_stats_timers, timer->seqnum)) is_stats_timer = TRUE; /* schedule immediatly expected timer which exceed the maximum RTX delay * reorder configuration */ if (priv->do_retransmission && priv->rtx_delay_reorder > 0) { RtpTimer *test = rtp_timer_queue_peek_earliest (priv->timers); while (test) { gint gap; /* filter the timer type to speed up this loop */ if (test->type != RTP_TIMER_EXPECTED) { test = rtp_timer_get_next (test); continue; } gap = gst_rtp_buffer_compare_seqnum (test->seqnum, seqnum); GST_DEBUG_OBJECT (jitterbuffer, "%d, #%d<->#%d gap %d", test->type, test->seqnum, seqnum, gap); /* if this expected packet have a smaller gap then the configured one, * then earlier timer are not expected to have bigger gap as the timer * queue is ordered */ if (gap <= priv->rtx_delay_reorder) break; /* max gap, we exceeded the max reorder distance and we don't expect the * missing packet to be this reordered */ if (test->num_rtx_retry == 0 && test->type == RTP_TIMER_EXPECTED) rtp_timer_queue_update_timer (priv->timers, test, test->seqnum, -1, 0, 0, FALSE); test = rtp_timer_get_next (test); } } do_next_seqnum = do_next_seqnum && priv->packet_spacing > 0 && priv->do_retransmission && priv->rtx_next_seqnum; if (timer && timer->type != RTP_TIMER_DEADLINE) { if (timer->num_rtx_retry > 0) { if (is_rtx) { update_rtx_stats (jitterbuffer, timer, dts, TRUE); /* don't try to estimate the next seqnum because this is a retransmitted * packet and it probably did not arrive with the expected packet * spacing. */ do_next_seqnum = FALSE; } if (!is_stats_timer && (!is_rtx || timer->num_rtx_retry > 1)) { RtpTimer *stats_timer = rtp_timer_dup (timer); /* Store timer in order to record stats when/if the retransmitted * packet arrives. We should also store timer information if we've * requested retransmission more than once since we may receive * several retransmitted packets. For accuracy we should update the * stats also when the redundant retransmitted packets arrives. */ stats_timer->timeout = pts + priv->rtx_stats_timeout * GST_MSECOND; stats_timer->type = RTP_TIMER_EXPECTED; rtp_timer_queue_insert (priv->rtx_stats_timers, stats_timer); } } } if (do_next_seqnum && pts != GST_CLOCK_TIME_NONE) { GstClockTime expected, delay; /* calculate expected arrival time of the next seqnum */ expected = pts + priv->packet_spacing; delay = get_rtx_delay (priv); /* and update/install timer for next seqnum */ GST_DEBUG_OBJECT (jitterbuffer, "Add RTX timer #%d, expected %" GST_TIME_FORMAT ", delay %" GST_TIME_FORMAT ", packet-spacing %" GST_TIME_FORMAT ", jitter %" GST_TIME_FORMAT, priv->next_in_seqnum, GST_TIME_ARGS (expected), GST_TIME_ARGS (delay), GST_TIME_ARGS (priv->packet_spacing), GST_TIME_ARGS (priv->avg_jitter)); if (timer && !is_stats_timer) { timer->type = RTP_TIMER_EXPECTED; rtp_timer_queue_update_timer (priv->timers, timer, priv->next_in_seqnum, expected, delay, 0, TRUE); } else { rtp_timer_queue_set_expected (priv->timers, priv->next_in_seqnum, expected, delay, priv->packet_spacing); } } else if (timer && timer->type != RTP_TIMER_DEADLINE && !is_stats_timer) { /* if we had a timer, remove it, we don't know when to expect the next * packet. */ rtp_timer_queue_unschedule (priv->timers, timer); rtp_timer_free (timer); } } static void calculate_packet_spacing (GstRtpJitterBuffer * jitterbuffer, guint32 rtptime, GstClockTime pts) { GstRtpJitterBufferPrivate *priv = jitterbuffer->priv; /* we need consecutive seqnums with a different * rtptime to estimate the packet spacing. */ if (priv->ips_rtptime != rtptime) { /* rtptime changed, check pts diff */ if (priv->ips_pts != -1 && pts != -1 && pts > priv->ips_pts) { GstClockTime new_packet_spacing = pts - priv->ips_pts; GstClockTime old_packet_spacing = priv->packet_spacing; /* Biased towards bigger packet spacings to prevent * too many unneeded retransmission requests for next * packets that just arrive a little later than we would * expect */ if (old_packet_spacing > new_packet_spacing) priv->packet_spacing = (new_packet_spacing + 3 * old_packet_spacing) / 4; else if (old_packet_spacing > 0) priv->packet_spacing = (3 * new_packet_spacing + old_packet_spacing) / 4; else priv->packet_spacing = new_packet_spacing; GST_DEBUG_OBJECT (jitterbuffer, "new packet spacing %" GST_TIME_FORMAT " old packet spacing %" GST_TIME_FORMAT " combined to %" GST_TIME_FORMAT, GST_TIME_ARGS (new_packet_spacing), GST_TIME_ARGS (old_packet_spacing), GST_TIME_ARGS (priv->packet_spacing)); } priv->ips_rtptime = rtptime; priv->ips_pts = pts; } } static void insert_lost_event (GstRtpJitterBuffer * jitterbuffer, guint16 seqnum, guint lost_packets, GstClockTime timestamp, GstClockTime duration, guint num_rtx_retry) { GstRtpJitterBufferPrivate *priv = jitterbuffer->priv; GstEvent *event = NULL; guint next_in_seqnum; /* we had a gap and thus we lost some packets. Create an event for this. */ if (lost_packets > 1) GST_DEBUG_OBJECT (jitterbuffer, "Packets #%d -> #%d lost", seqnum, seqnum + lost_packets - 1); else GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d lost", seqnum); priv->num_lost += lost_packets; priv->num_rtx_failed += num_rtx_retry; next_in_seqnum = (seqnum + lost_packets) & 0xffff; /* we now only accept seqnum bigger than this */ if (gst_rtp_buffer_compare_seqnum (priv->next_in_seqnum, next_in_seqnum) > 0) { priv->next_in_seqnum = next_in_seqnum; priv->last_in_pts = timestamp; } /* Avoid creating events if we don't need it. Note that we still need to create * the lost *ITEM* since it will be used to notify the outgoing thread of * lost items (so that we can set discont flags and such) */ if (priv->do_lost) { /* create packet lost event */ if (duration == GST_CLOCK_TIME_NONE && priv->packet_spacing > 0) duration = priv->packet_spacing; event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM, gst_structure_new ("GstRTPPacketLost", "seqnum", G_TYPE_UINT, (guint) seqnum, "timestamp", G_TYPE_UINT64, timestamp, "duration", G_TYPE_UINT64, duration, "retry", G_TYPE_UINT, num_rtx_retry, NULL)); } if (rtp_jitter_buffer_append_lost_event (priv->jbuf, event, seqnum, lost_packets)) JBUF_SIGNAL_EVENT (priv); } static void calculate_expected (GstRtpJitterBuffer * jitterbuffer, guint32 expected, guint16 seqnum, GstClockTime pts, gint gap) { GstRtpJitterBufferPrivate *priv = jitterbuffer->priv; GstClockTime duration, expected_pts; gboolean equidistant = priv->equidistant > 0; GstClockTime last_in_pts = priv->last_in_pts; GST_DEBUG_OBJECT (jitterbuffer, "pts %" GST_TIME_FORMAT ", last %" GST_TIME_FORMAT, GST_TIME_ARGS (pts), GST_TIME_ARGS (last_in_pts)); if (pts == GST_CLOCK_TIME_NONE) { GST_WARNING_OBJECT (jitterbuffer, "Have no PTS"); return; } if (equidistant) { GstClockTime total_duration; /* the total duration spanned by the missing packets */ if (pts >= last_in_pts) total_duration = pts - last_in_pts; else total_duration = 0; /* 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. */ duration = total_duration / (gap + 1); GST_DEBUG_OBJECT (jitterbuffer, "duration %" GST_TIME_FORMAT, GST_TIME_ARGS (duration)); if (total_duration > priv->latency_ns) { GstClockTime gap_time; guint lost_packets; if (duration > 0) { GstClockTime gap_dur = gap * duration; if (gap_dur > priv->latency_ns) gap_time = gap_dur - priv->latency_ns; else gap_time = 0; lost_packets = gap_time / duration; } else { gap_time = total_duration - priv->latency_ns; lost_packets = gap; } /* too many lost packets, some of the missing packets are already * too late and we can generate lost packet events for them. */ GST_INFO_OBJECT (jitterbuffer, "lost packets (%d, #%d->#%d) duration too large %" GST_TIME_FORMAT " > %" GST_TIME_FORMAT ", consider %u lost (%" GST_TIME_FORMAT ")", gap, expected, seqnum - 1, GST_TIME_ARGS (total_duration), GST_TIME_ARGS (priv->latency_ns), lost_packets, GST_TIME_ARGS (gap_time)); /* this multi-lost-packet event will be inserted directly into the packet-queue for immediate processing */ if (lost_packets > 0) { RtpTimer *timer; GstClockTime timestamp = apply_offset (jitterbuffer, last_in_pts + duration); insert_lost_event (jitterbuffer, expected, lost_packets, timestamp, gap_time, 0); timer = rtp_timer_queue_find (priv->timers, expected); if (timer && timer->type == RTP_TIMER_EXPECTED) { if (timer->queued) rtp_timer_queue_unschedule (priv->timers, timer); GST_DEBUG_OBJECT (jitterbuffer, "removing timer for seqnum #%u", expected); rtp_timer_free (timer); } expected += lost_packets; last_in_pts += gap_time; } } expected_pts = last_in_pts + duration; } else { /* If we cannot assume equidistant packet spacing, the only thing we now * for sure is that the missing packets have expected pts not later than * the last received pts. */ duration = 0; expected_pts = pts; } if (priv->do_retransmission) { RtpTimer *timer = rtp_timer_queue_find (priv->timers, expected); GstClockTime rtx_delay = get_rtx_delay (priv); /* if we had a timer for the first missing packet, update it. */ if (timer && timer->type == RTP_TIMER_EXPECTED) { GstClockTime timeout = timer->timeout; GstClockTime delay = MAX (rtx_delay, pts - expected_pts); timer->duration = duration; if (timeout > (expected_pts + delay) && timer->num_rtx_retry == 0) { rtp_timer_queue_update_timer (priv->timers, timer, timer->seqnum, expected_pts, delay, 0, TRUE); } expected++; expected_pts += duration; } while (gst_rtp_buffer_compare_seqnum (expected, seqnum) > 0) { /* minimum delay the expected-timer has "waited" is the elapsed time * since expected arrival of the missing packet */ GstClockTime delay = MAX (rtx_delay, pts - expected_pts); rtp_timer_queue_set_expected (priv->timers, expected, expected_pts, delay, duration); expected_pts += duration; expected++; } } else { while (gst_rtp_buffer_compare_seqnum (expected, seqnum) > 0) { rtp_timer_queue_set_lost (priv->timers, expected, expected_pts, duration, timeout_offset (jitterbuffer)); expected_pts += duration; expected++; } } } static void calculate_jitter (GstRtpJitterBuffer * jitterbuffer, GstClockTime dts, guint32 rtptime) { gint32 rtpdiff; GstClockTimeDiff dtsdiff, rtpdiffns, diff; GstRtpJitterBufferPrivate *priv; priv = jitterbuffer->priv; if (G_UNLIKELY (dts == GST_CLOCK_TIME_NONE) || priv->clock_rate <= 0) goto no_time; if (priv->last_dts != -1) dtsdiff = dts - priv->last_dts; else dtsdiff = 0; if (priv->last_rtptime != -1) rtpdiff = rtptime - (guint32) priv->last_rtptime; else rtpdiff = 0; /* Guess whether stream currently uses equidistant packet spacing. If we * often see identical timestamps it means the packets are not * equidistant. */ if (rtptime == priv->last_rtptime) priv->equidistant -= 2; else priv->equidistant += 1; priv->equidistant = CLAMP (priv->equidistant, -7, 7); priv->last_dts = dts; priv->last_rtptime = rtptime; if (rtpdiff > 0) rtpdiffns = gst_util_uint64_scale_int (rtpdiff, GST_SECOND, priv->clock_rate); else rtpdiffns = -gst_util_uint64_scale_int (-rtpdiff, GST_SECOND, priv->clock_rate); diff = ABS (dtsdiff - rtpdiffns); /* jitter is stored in nanoseconds */ priv->avg_jitter = (diff + (15 * priv->avg_jitter)) >> 4; GST_LOG_OBJECT (jitterbuffer, "dtsdiff %" GST_STIME_FORMAT " rtptime %" GST_STIME_FORMAT ", clock-rate %d, diff %" GST_STIME_FORMAT ", jitter: %" GST_TIME_FORMAT, GST_STIME_ARGS (dtsdiff), GST_STIME_ARGS (rtpdiffns), priv->clock_rate, GST_STIME_ARGS (diff), GST_TIME_ARGS (priv->avg_jitter)); return; /* ERRORS */ no_time: { GST_DEBUG_OBJECT (jitterbuffer, "no dts or no clock-rate, can't calculate jitter"); return; } } static gint compare_buffer_seqnum (GstBuffer * a, GstBuffer * b, gpointer user_data) { GstRTPBuffer rtp_a = GST_RTP_BUFFER_INIT; GstRTPBuffer rtp_b = GST_RTP_BUFFER_INIT; guint seq_a, seq_b; gst_rtp_buffer_map (a, GST_MAP_READ, &rtp_a); seq_a = gst_rtp_buffer_get_seq (&rtp_a); gst_rtp_buffer_unmap (&rtp_a); gst_rtp_buffer_map (b, GST_MAP_READ, &rtp_b); seq_b = gst_rtp_buffer_get_seq (&rtp_b); gst_rtp_buffer_unmap (&rtp_b); return gst_rtp_buffer_compare_seqnum (seq_b, seq_a); } static gboolean handle_big_gap_buffer (GstRtpJitterBuffer * jitterbuffer, GstBuffer * buffer, guint8 pt, guint16 seqnum, gint gap, guint max_dropout, guint max_misorder) { GstRtpJitterBufferPrivate *priv; guint gap_packets_length; gboolean reset = FALSE; gboolean future = gap > 0; priv = jitterbuffer->priv; if ((gap_packets_length = g_queue_get_length (&priv->gap_packets)) > 0) { GList *l; guint32 prev_gap_seq = -1; gboolean all_consecutive = TRUE; g_queue_insert_sorted (&priv->gap_packets, buffer, (GCompareDataFunc) compare_buffer_seqnum, NULL); for (l = priv->gap_packets.head; l; l = l->next) { GstBuffer *gap_buffer = l->data; GstRTPBuffer gap_rtp = GST_RTP_BUFFER_INIT; guint32 gap_seq; gst_rtp_buffer_map (gap_buffer, GST_MAP_READ, &gap_rtp); all_consecutive = (gst_rtp_buffer_get_payload_type (&gap_rtp) == pt); gap_seq = gst_rtp_buffer_get_seq (&gap_rtp); if (prev_gap_seq == -1) prev_gap_seq = gap_seq; else if (gst_rtp_buffer_compare_seqnum (gap_seq, prev_gap_seq) != -1) all_consecutive = FALSE; else prev_gap_seq = gap_seq; gst_rtp_buffer_unmap (&gap_rtp); if (!all_consecutive) break; } if (all_consecutive && gap_packets_length > 3) { GST_DEBUG_OBJECT (jitterbuffer, "buffer too %s %d < %d, got 5 consecutive ones - reset", (future ? "new" : "old"), gap, (future ? max_dropout : -max_misorder)); reset = TRUE; } else if (!all_consecutive) { g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL); g_queue_clear (&priv->gap_packets); GST_DEBUG_OBJECT (jitterbuffer, "buffer too %s %d < %d, got no 5 consecutive ones - dropping", (future ? "new" : "old"), gap, (future ? max_dropout : -max_misorder)); buffer = NULL; } else { GST_DEBUG_OBJECT (jitterbuffer, "buffer too %s %d < %d, got %u consecutive ones - waiting", (future ? "new" : "old"), gap, (future ? max_dropout : -max_misorder), gap_packets_length + 1); buffer = NULL; } } else { GST_DEBUG_OBJECT (jitterbuffer, "buffer too %s %d < %d, first one - waiting", (future ? "new" : "old"), gap, -max_misorder); g_queue_push_tail (&priv->gap_packets, buffer); buffer = NULL; } return reset; } static GstClockTime get_current_running_time (GstRtpJitterBuffer * jitterbuffer) { GstClock *clock = gst_element_get_clock (GST_ELEMENT_CAST (jitterbuffer)); GstClockTime running_time = GST_CLOCK_TIME_NONE; if (clock) { GstClockTime base_time = gst_element_get_base_time (GST_ELEMENT_CAST (jitterbuffer)); GstClockTime clock_time = gst_clock_get_time (clock); if (clock_time > base_time) running_time = clock_time - base_time; else running_time = 0; gst_object_unref (clock); } return running_time; } static GstFlowReturn gst_rtp_jitter_buffer_reset (GstRtpJitterBuffer * jitterbuffer, GstPad * pad, GstObject * parent, guint16 seqnum) { GstRtpJitterBufferPrivate *priv = jitterbuffer->priv; GstFlowReturn ret = GST_FLOW_OK; GList *events = NULL, *l; GList *buffers; GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer"); rtp_jitter_buffer_flush (priv->jbuf, (GFunc) free_item_and_retain_sticky_events, &events); rtp_jitter_buffer_reset_skew (priv->jbuf); rtp_timer_queue_remove_all (priv->timers); priv->discont = TRUE; priv->last_popped_seqnum = -1; if (priv->gap_packets.head) { GstBuffer *gap_buffer = priv->gap_packets.head->data; GstRTPBuffer gap_rtp = GST_RTP_BUFFER_INIT; gst_rtp_buffer_map (gap_buffer, GST_MAP_READ, &gap_rtp); priv->next_seqnum = gst_rtp_buffer_get_seq (&gap_rtp); gst_rtp_buffer_unmap (&gap_rtp); } else { priv->next_seqnum = seqnum; } priv->last_in_pts = -1; priv->next_in_seqnum = -1; /* Insert all sticky events again in order, otherwise we would * potentially loose STREAM_START, CAPS or SEGMENT events */ events = g_list_reverse (events); for (l = events; l; l = l->next) { rtp_jitter_buffer_append_event (priv->jbuf, l->data); } g_list_free (events); JBUF_SIGNAL_EVENT (priv); /* reset spacing estimation when gap */ priv->ips_rtptime = -1; priv->ips_pts = GST_CLOCK_TIME_NONE; buffers = g_list_copy (priv->gap_packets.head); g_queue_clear (&priv->gap_packets); priv->ips_rtptime = -1; priv->ips_pts = GST_CLOCK_TIME_NONE; JBUF_UNLOCK (jitterbuffer->priv); for (l = buffers; l; l = l->next) { ret = gst_rtp_jitter_buffer_chain (pad, parent, l->data); l->data = NULL; if (ret != GST_FLOW_OK) { l = l->next; break; } } for (; l; l = l->next) gst_buffer_unref (l->data); g_list_free (buffers); return ret; } static gboolean gst_rtp_jitter_buffer_fast_start (GstRtpJitterBuffer * jitterbuffer) { GstRtpJitterBufferPrivate *priv; RTPJitterBufferItem *item; RtpTimer *timer; priv = jitterbuffer->priv; if (priv->faststart_min_packets == 0) return FALSE; item = rtp_jitter_buffer_peek (priv->jbuf); if (!item) return FALSE; timer = rtp_timer_queue_find (priv->timers, item->seqnum); if (!timer || timer->type != RTP_TIMER_DEADLINE) return FALSE; if (rtp_jitter_buffer_can_fast_start (priv->jbuf, priv->faststart_min_packets)) { GST_INFO_OBJECT (jitterbuffer, "We found %i consecutive packet, start now", priv->faststart_min_packets); timer->timeout = -1; rtp_timer_queue_reschedule (priv->timers, timer); return TRUE; } return FALSE; } static GstFlowReturn gst_rtp_jitter_buffer_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer) { GstRtpJitterBuffer *jitterbuffer; GstRtpJitterBufferPrivate *priv; guint16 seqnum; guint32 expected, rtptime; GstFlowReturn ret = GST_FLOW_OK; GstClockTime dts, pts; guint64 latency_ts; gboolean head; gboolean duplicate; gint percent = -1; guint8 pt; GstRTPBuffer rtp = GST_RTP_BUFFER_INIT; gboolean do_next_seqnum = FALSE; GstMessage *msg = NULL; GstMessage *drop_msg = NULL; gboolean estimated_dts = FALSE; gint32 packet_rate, max_dropout, max_misorder; RtpTimer *timer = NULL; gboolean is_rtx; jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent); priv = jitterbuffer->priv; if (G_UNLIKELY (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp))) goto invalid_buffer; pt = gst_rtp_buffer_get_payload_type (&rtp); seqnum = gst_rtp_buffer_get_seq (&rtp); rtptime = gst_rtp_buffer_get_timestamp (&rtp); gst_rtp_buffer_unmap (&rtp); is_rtx = GST_BUFFER_IS_RETRANSMISSION (buffer); /* make sure we have PTS and DTS set */ pts = GST_BUFFER_PTS (buffer); dts = GST_BUFFER_DTS (buffer); if (dts == -1) dts = pts; else if (pts == -1) pts = dts; if (dts == -1) { /* If we have no DTS here, i.e. no capture time, get one from the * clock now to have something to calculate with in the future. */ dts = get_current_running_time (jitterbuffer); pts = dts; /* Remember that we estimated the DTS if we are running already * and this is not our first packet (or first packet after a reset). * If it's the first packet, we somehow must generate a timestamp for * everything, otherwise we can't calculate any times */ estimated_dts = (priv->next_in_seqnum != -1); } else { /* take the DTS 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 DTS with the smoothed value after processing it in the * jitterbuffer and assign it as the PTS. */ /* bring to running time */ dts = gst_segment_to_running_time (&priv->segment, GST_FORMAT_TIME, dts); } GST_DEBUG_OBJECT (jitterbuffer, "Received packet #%d at time %" GST_TIME_FORMAT ", discont %d, rtx %d", seqnum, GST_TIME_ARGS (dts), GST_BUFFER_IS_DISCONT (buffer), is_rtx); 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_pad_get_current_caps (pad))) { gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps, pt); gst_caps_unref (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_FLUSHING) goto out_flushing; if (G_UNLIKELY (priv->clock_rate == -1)) goto no_clock_rate; gst_rtp_packet_rate_ctx_reset (&priv->packet_rate_ctx, priv->clock_rate); } /* don't accept more data on EOS */ if (G_UNLIKELY (priv->eos)) goto have_eos; if (!is_rtx) calculate_jitter (jitterbuffer, dts, rtptime); if (priv->seqnum_base != -1) { gint gap; gap = gst_rtp_buffer_compare_seqnum (priv->seqnum_base, seqnum); if (gap < 0) { GST_DEBUG_OBJECT (jitterbuffer, "packet seqnum #%d before seqnum-base #%d", seqnum, priv->seqnum_base); gst_buffer_unref (buffer); goto finished; } else if (gap > 16384) { /* From now on don't compare against the seqnum base anymore as * at some point in the future we will wrap around and also that * much reordering is very unlikely */ priv->seqnum_base = -1; } } expected = priv->next_in_seqnum; /* don't update packet-rate based on RTX, as those arrive highly unregularly */ if (!is_rtx) { packet_rate = gst_rtp_packet_rate_ctx_update (&priv->packet_rate_ctx, seqnum, rtptime); GST_TRACE_OBJECT (jitterbuffer, "updated packet_rate: %d", packet_rate); } max_dropout = gst_rtp_packet_rate_ctx_get_max_dropout (&priv->packet_rate_ctx, priv->max_dropout_time); max_misorder = gst_rtp_packet_rate_ctx_get_max_misorder (&priv->packet_rate_ctx, priv->max_misorder_time); GST_TRACE_OBJECT (jitterbuffer, "max_dropout: %d, max_misorder: %d", max_dropout, max_misorder); timer = rtp_timer_queue_find (priv->timers, seqnum); if (is_rtx) { if (G_UNLIKELY (!priv->do_retransmission)) goto unsolicited_rtx; if (!timer) timer = rtp_timer_queue_find (priv->rtx_stats_timers, seqnum); /* If the first buffer is an (old) rtx, e.g. from before a reset, or * already lost, ignore it */ if (!timer || expected == -1) goto unsolicited_rtx; } /* now check against our expected seqnum */ if (G_UNLIKELY (expected == -1)) { GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum); /* calculate a pts based on rtptime and arrival time (dts) */ pts = rtp_jitter_buffer_calculate_pts (priv->jbuf, dts, estimated_dts, rtptime, gst_element_get_base_time (GST_ELEMENT_CAST (jitterbuffer)), 0, FALSE); if (G_UNLIKELY (!GST_CLOCK_TIME_IS_VALID (pts))) { /* A valid timestamp cannot be calculated, discard packet */ goto discard_invalid; } /* we don't know what the next_in_seqnum should be, wait for the last * possible moment to push this buffer, maybe we get an earlier seqnum * while we wait */ rtp_timer_queue_set_deadline (priv->timers, seqnum, pts, timeout_offset (jitterbuffer)); do_next_seqnum = TRUE; /* take rtptime and pts to calculate packet spacing */ priv->ips_rtptime = rtptime; priv->ips_pts = pts; } else { gint gap; /* now calculate gap */ gap = gst_rtp_buffer_compare_seqnum (expected, seqnum); GST_DEBUG_OBJECT (jitterbuffer, "expected #%d, got #%d, gap of %d", expected, seqnum, gap); if (G_UNLIKELY (gap > 0 && rtp_timer_queue_length (priv->timers) >= max_dropout)) { /* If we have timers for more than RTP_MAX_DROPOUT packets * pending this means that we have a huge gap overall. We can * reset the jitterbuffer at this point because there's * just too much data missing to be able to do anything * sensible with the past data. Just try again from the * next packet */ GST_WARNING_OBJECT (jitterbuffer, "%d pending timers > %d - resetting", rtp_timer_queue_length (priv->timers), max_dropout); g_queue_insert_sorted (&priv->gap_packets, buffer, (GCompareDataFunc) compare_buffer_seqnum, NULL); return gst_rtp_jitter_buffer_reset (jitterbuffer, pad, parent, seqnum); } /* Special handling of large gaps */ if (!is_rtx && ((gap != -1 && gap < -max_misorder) || (gap >= max_dropout))) { gboolean reset = handle_big_gap_buffer (jitterbuffer, buffer, pt, seqnum, gap, max_dropout, max_misorder); if (reset) { return gst_rtp_jitter_buffer_reset (jitterbuffer, pad, parent, seqnum); } else { GST_DEBUG_OBJECT (jitterbuffer, "Had big gap, waiting for more consecutive packets"); goto finished; } } /* We had no huge gap, let's drop all the gap packets */ GST_DEBUG_OBJECT (jitterbuffer, "Clearing gap packets"); g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL); g_queue_clear (&priv->gap_packets); /* calculate a pts based on rtptime and arrival time (dts) */ /* If we estimated the DTS, don't consider it in the clock skew calculations */ pts = rtp_jitter_buffer_calculate_pts (priv->jbuf, dts, estimated_dts, rtptime, gst_element_get_base_time (GST_ELEMENT_CAST (jitterbuffer)), gap, is_rtx); if (G_UNLIKELY (!GST_CLOCK_TIME_IS_VALID (pts))) { /* A valid timestamp cannot be calculated, discard packet */ goto discard_invalid; } if (G_LIKELY (gap == 0)) { /* packet is expected */ calculate_packet_spacing (jitterbuffer, rtptime, pts); do_next_seqnum = TRUE; } else { /* we have a gap */ if (gap > 0) { GST_DEBUG_OBJECT (jitterbuffer, "%d missing packets", gap); /* fill in the gap with EXPECTED timers */ calculate_expected (jitterbuffer, expected, seqnum, pts, gap); do_next_seqnum = TRUE; } else { GST_DEBUG_OBJECT (jitterbuffer, "old packet received"); do_next_seqnum = FALSE; } /* reset spacing estimation when gap */ priv->ips_rtptime = -1; priv->ips_pts = GST_CLOCK_TIME_NONE; } } if (do_next_seqnum) { priv->last_in_pts = pts; priv->next_in_seqnum = (seqnum + 1) & 0xffff; } if (is_rtx) timer->num_rtx_received++; /* At 2^15, we would detect a seqnum rollover too early, therefore * limit the queue size. But let's not limit it to a number that is * too small to avoid emptying it needlessly if there is a spurious huge * sequence number, let's allow at least 10k packets in any case. */ while (rtp_jitter_buffer_is_full (priv->jbuf) && priv->srcresult == GST_FLOW_OK) { RtpTimer *timer = rtp_timer_queue_peek_earliest (priv->timers); while (timer) { timer->timeout = -1; if (timer->type == RTP_TIMER_DEADLINE) break; timer = rtp_timer_get_next (timer); } update_current_timer (jitterbuffer); JBUF_WAIT_QUEUE (priv); if (priv->srcresult != GST_FLOW_OK) goto out_flushing; } /* 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)) { if (priv->do_retransmission) { if (is_rtx && timer) { update_rtx_stats (jitterbuffer, timer, dts, FALSE); /* Only count the retranmitted packet too late if it has been * considered lost. If the original packet arrived before the * retransmitted we just count it as a duplicate. */ if (timer->type != RTP_TIMER_LOST) goto rtx_duplicate; } } 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)) { RTPJitterBufferItem *old_item; old_item = rtp_jitter_buffer_peek (priv->jbuf); if (IS_DROPABLE (old_item)) { old_item = rtp_jitter_buffer_pop (priv->jbuf, &percent); GST_DEBUG_OBJECT (jitterbuffer, "Queue full, dropping old packet %p", old_item); priv->next_seqnum = (old_item->seqnum + old_item->count) & 0xffff; if (priv->post_drop_messages) { drop_msg = new_drop_message (jitterbuffer, old_item->seqnum, old_item->pts, REASON_DROP_ON_LATENCY); } rtp_jitter_buffer_free_item (old_item); } /* we might have removed some head buffers, signal the pushing thread to * see if it can push now */ JBUF_SIGNAL_EVENT (priv); } } /* If we estimated the DTS, don't consider it in the clock skew calculations * later. The code above always sets dts to pts or the other way around if * any of those is valid in the buffer, so we know that if we estimated the * dts that both are unknown */ head = rtp_jitter_buffer_append_buffer (priv->jbuf, buffer, estimated_dts ? GST_CLOCK_TIME_NONE : dts, pts, seqnum, rtptime, &duplicate, &percent); /* 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 (duplicate)) { if (is_rtx && timer) update_rtx_stats (jitterbuffer, timer, dts, FALSE); goto duplicate; } /* Trigger fast start if needed */ if (gst_rtp_jitter_buffer_fast_start (jitterbuffer)) head = TRUE; /* update timers */ update_timers (jitterbuffer, seqnum, dts, pts, do_next_seqnum, is_rtx, timer); /* we had an unhandled SR, handle it now */ if (priv->last_sr) do_handle_sync (jitterbuffer); if (G_UNLIKELY (head)) { /* signal addition of new buffer when the _loop is waiting. */ if (G_LIKELY (priv->active)) JBUF_SIGNAL_EVENT (priv); } GST_DEBUG_OBJECT (jitterbuffer, "Pushed packet #%d, now %d packets, head: %d, " "percent %d", seqnum, rtp_jitter_buffer_num_packets (priv->jbuf), head, percent); msg = check_buffering_percent (jitterbuffer, percent); finished: update_current_timer (jitterbuffer); JBUF_UNLOCK (priv); if (msg) gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), msg); if (drop_msg) gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), drop_msg); 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); 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_EOS; GST_WARNING_OBJECT (jitterbuffer, "we are EOS, refusing buffer"); gst_buffer_unref (buffer); goto finished; } too_late: { GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d too late as #%d was already" " popped, dropping", seqnum, priv->last_popped_seqnum); priv->num_late++; if (priv->post_drop_messages) { drop_msg = new_drop_message (jitterbuffer, seqnum, pts, REASON_TOO_LATE); } gst_buffer_unref (buffer); goto finished; } duplicate: { GST_DEBUG_OBJECT (jitterbuffer, "Duplicate packet #%d detected, dropping", seqnum); priv->num_duplicates++; goto finished; } rtx_duplicate: { GST_DEBUG_OBJECT (jitterbuffer, "Duplicate RTX packet #%d detected, dropping", seqnum); priv->num_duplicates++; gst_buffer_unref (buffer); goto finished; } unsolicited_rtx: { GST_DEBUG_OBJECT (jitterbuffer, "Unsolicited RTX packet #%d detected, dropping", seqnum); gst_buffer_unref (buffer); goto finished; } discard_invalid: { GST_DEBUG_OBJECT (jitterbuffer, "cannot calculate a valid pts for #%d (rtx: %d), discard", seqnum, is_rtx); gst_buffer_unref (buffer); goto finished; } } /* FIXME: hopefully we can do something more efficient here, especially when * all packets are in order and/or outside of the currently cached range. * Still worthwhile to have it, avoids taking/releasing object lock and pad * stream lock for every single buffer in the default chain_list fallback. */ static GstFlowReturn gst_rtp_jitter_buffer_chain_list (GstPad * pad, GstObject * parent, GstBufferList * buffer_list) { GstFlowReturn flow_ret = GST_FLOW_OK; guint i, n; n = gst_buffer_list_length (buffer_list); for (i = 0; i < n; ++i) { GstBuffer *buf = gst_buffer_list_get (buffer_list, i); flow_ret = gst_rtp_jitter_buffer_chain (pad, parent, gst_buffer_ref (buf)); if (flow_ret != GST_FLOW_OK) break; } gst_buffer_list_unref (buffer_list); return flow_ret; } static GstClockTime compute_elapsed (GstRtpJitterBuffer * jitterbuffer, RTPJitterBufferItem * item) { guint64 ext_time, elapsed; guint32 rtp_time; GstRtpJitterBufferPrivate *priv; priv = jitterbuffer->priv; rtp_time = item->rtptime; GST_LOG_OBJECT (jitterbuffer, "rtp %" G_GUINT32_FORMAT ", ext %" G_GUINT64_FORMAT, rtp_time, priv->ext_timestamp); ext_time = priv->ext_timestamp; ext_time = gst_rtp_buffer_ext_timestamp (&ext_time, rtp_time); if (ext_time < priv->ext_timestamp) { ext_time = priv->ext_timestamp; } else { priv->ext_timestamp = ext_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; } static void update_estimated_eos (GstRtpJitterBuffer * jitterbuffer, RTPJitterBufferItem * item) { guint64 total, elapsed, left, estimated; GstClockTime out_time; GstRtpJitterBufferPrivate *priv = jitterbuffer->priv; if (priv->npt_stop == -1 || priv->ext_timestamp == -1 || priv->clock_base == -1 || priv->clock_rate <= 0) return; /* compute the elapsed time */ elapsed = compute_elapsed (jitterbuffer, item); /* do nothing if elapsed time doesn't increment */ if (priv->last_elapsed && elapsed <= priv->last_elapsed) return; priv->last_elapsed = elapsed; /* this is the total time we need to play */ total = priv->npt_stop - priv->npt_start; GST_LOG_OBJECT (jitterbuffer, "total %" GST_TIME_FORMAT, GST_TIME_ARGS (total)); /* this is how much time there is left */ if (total > elapsed) left = total - elapsed; else left = 0; /* if we have less time left that the size of the buffer, we will not * be able to keep it filled, disabled buffering then */ if (left < rtp_jitter_buffer_get_delay (priv->jbuf)) { GST_DEBUG_OBJECT (jitterbuffer, "left %" GST_TIME_FORMAT ", disable buffering close to EOS", GST_TIME_ARGS (left)); rtp_jitter_buffer_disable_buffering (priv->jbuf, TRUE); } /* this is the current time as running-time */ out_time = item->pts; if (elapsed > 0) estimated = gst_util_uint64_scale (out_time, total, elapsed); else { /* if there is almost nothing left, * we may never advance enough to end up in the above case */ if (total < 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)); if (estimated != -1 && priv->estimated_eos != estimated) { rtp_timer_queue_set_eos (priv->timers, estimated, timeout_offset (jitterbuffer)); priv->estimated_eos = estimated; } } /* take a buffer from the queue and push it */ static GstFlowReturn pop_and_push_next (GstRtpJitterBuffer * jitterbuffer, guint seqnum) { GstRtpJitterBufferPrivate *priv = jitterbuffer->priv; GstFlowReturn result = GST_FLOW_OK; RTPJitterBufferItem *item; GstBuffer *outbuf = NULL; GstEvent *outevent = NULL; GstQuery *outquery = NULL; GstClockTime dts, pts; gint percent = -1; gboolean do_push = TRUE; guint type; GstMessage *msg; /* when we get here we are ready to pop and push the buffer */ item = rtp_jitter_buffer_pop (priv->jbuf, &percent); type = item->type; switch (type) { case ITEM_TYPE_BUFFER: /* we need to make writable to change the flags and timestamps */ outbuf = gst_buffer_make_writable (item->data); if (G_UNLIKELY (priv->discont)) { /* set DISCONT flag when we missed a packet. We pushed the buffer writable * into the jitterbuffer so we can modify now. */ GST_DEBUG_OBJECT (jitterbuffer, "mark output buffer discont"); GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT); priv->discont = FALSE; } if (G_UNLIKELY (priv->ts_discont)) { GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC); priv->ts_discont = FALSE; } dts = gst_segment_position_from_running_time (&priv->segment, GST_FORMAT_TIME, item->dts); pts = gst_segment_position_from_running_time (&priv->segment, GST_FORMAT_TIME, item->pts); /* if this is a new frame, check if ts_offset needs to be updated */ if (pts != priv->last_pts) { update_offset (jitterbuffer); } /* apply timestamp with offset to buffer now */ GST_BUFFER_DTS (outbuf) = apply_offset (jitterbuffer, dts); GST_BUFFER_PTS (outbuf) = apply_offset (jitterbuffer, pts); /* update the elapsed time when we need to check against the npt stop time. */ update_estimated_eos (jitterbuffer, item); priv->last_pts = pts; priv->last_out_time = GST_BUFFER_PTS (outbuf); break; case ITEM_TYPE_LOST: priv->discont = TRUE; if (!priv->do_lost) do_push = FALSE; /* FALLTHROUGH */ case ITEM_TYPE_EVENT: outevent = item->data; break; case ITEM_TYPE_QUERY: outquery = item->data; break; } /* 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. */ if (seqnum != -1) { priv->last_popped_seqnum = seqnum; priv->next_seqnum = (seqnum + item->count) & 0xffff; } msg = check_buffering_percent (jitterbuffer, percent); if (type == ITEM_TYPE_EVENT && outevent && GST_EVENT_TYPE (outevent) == GST_EVENT_EOS) { g_assert (priv->eos); while (rtp_timer_queue_length (priv->timers) > 0) { /* Stopping timers */ unschedule_current_timer (jitterbuffer); JBUF_WAIT_TIMER (priv); } } JBUF_UNLOCK (priv); item->data = NULL; rtp_jitter_buffer_free_item (item); if (msg) gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), msg); switch (type) { case ITEM_TYPE_BUFFER: /* push buffer */ GST_DEBUG_OBJECT (jitterbuffer, "Pushing buffer %d, dts %" GST_TIME_FORMAT ", pts %" GST_TIME_FORMAT, seqnum, GST_TIME_ARGS (GST_BUFFER_DTS (outbuf)), GST_TIME_ARGS (GST_BUFFER_PTS (outbuf))); priv->num_pushed++; GST_BUFFER_DTS (outbuf) = GST_CLOCK_TIME_NONE; result = gst_pad_push (priv->srcpad, outbuf); JBUF_LOCK_CHECK (priv, out_flushing); break; case ITEM_TYPE_LOST: case ITEM_TYPE_EVENT: /* We got not enough consecutive packets with a huge gap, we can * as well just drop them here now on EOS */ if (outevent && GST_EVENT_TYPE (outevent) == GST_EVENT_EOS) { GST_DEBUG_OBJECT (jitterbuffer, "Clearing gap packets on EOS"); g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL); g_queue_clear (&priv->gap_packets); } GST_DEBUG_OBJECT (jitterbuffer, "%sPushing event %" GST_PTR_FORMAT ", seqnum %d", do_push ? "" : "NOT ", outevent, seqnum); if (do_push) gst_pad_push_event (priv->srcpad, outevent); else if (outevent) gst_event_unref (outevent); result = GST_FLOW_OK; JBUF_LOCK_CHECK (priv, out_flushing); break; case ITEM_TYPE_QUERY: { gboolean res; res = gst_pad_peer_query (priv->srcpad, outquery); JBUF_LOCK_CHECK (priv, out_flushing); result = GST_FLOW_OK; GST_LOG_OBJECT (jitterbuffer, "did query %p, return %d", outquery, res); JBUF_SIGNAL_QUERY (priv, res); break; } } return result; /* ERRORS */ out_flushing: { return priv->srcresult; } } #define GST_FLOW_WAIT GST_FLOW_CUSTOM_SUCCESS /* Peek a buffer and compare the seqnum to the expected seqnum. * If all is fine, the buffer is pushed. * If something is wrong, we wait for some event */ static GstFlowReturn handle_next_buffer (GstRtpJitterBuffer * jitterbuffer) { GstRtpJitterBufferPrivate *priv = jitterbuffer->priv; GstFlowReturn result; RTPJitterBufferItem *item; guint seqnum; guint32 next_seqnum; /* only push buffers when PLAYING and active and not buffering */ if (priv->blocked || !priv->active || rtp_jitter_buffer_is_buffering (priv->jbuf)) { return GST_FLOW_WAIT; } /* peek a buffer, we're just looking at the sequence number. * If all is fine, we'll pop and push it. If the sequence number is wrong we * wait for a timeout or something to change. * The peeked buffer is valid for as long as we hold the jitterbuffer lock. */ item = rtp_jitter_buffer_peek (priv->jbuf); if (item == NULL) { goto wait; } /* get the seqnum and the next expected seqnum */ seqnum = item->seqnum; if (seqnum == -1) { return pop_and_push_next (jitterbuffer, seqnum); } next_seqnum = priv->next_seqnum; /* get the gap between this and the previous packet. If we don't know the * previous packet seqnum assume no gap. */ if (G_UNLIKELY (next_seqnum == -1)) { GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum); /* we don't know what the next_seqnum should be, the chain function should * have scheduled a DEADLINE timer that will increment next_seqnum when it * fires, so wait for that */ result = GST_FLOW_WAIT; } else { gint gap = gst_rtp_buffer_compare_seqnum (next_seqnum, seqnum); if (G_LIKELY (gap == 0)) { /* no missing packet, pop and push */ result = pop_and_push_next (jitterbuffer, seqnum); } else if (G_UNLIKELY (gap < 0)) { /* if we have a packet that we already pushed or considered dropped, pop it * off and get the next packet */ GST_DEBUG_OBJECT (jitterbuffer, "Old packet #%d, next #%d dropping", seqnum, next_seqnum); item = rtp_jitter_buffer_pop (priv->jbuf, NULL); rtp_jitter_buffer_free_item (item); result = GST_FLOW_OK; } else { /* the chain function has scheduled timers to request retransmission or * when to consider the packet lost, wait for that */ GST_DEBUG_OBJECT (jitterbuffer, "Sequence number GAP detected: expected %d instead of %d (%d missing)", next_seqnum, seqnum, gap); /* if we have reached EOS, just keep processing */ /* Also do the same if we block input because the JB is full */ if (priv->eos || rtp_jitter_buffer_is_full (priv->jbuf)) { result = pop_and_push_next (jitterbuffer, seqnum); result = GST_FLOW_OK; } else { result = GST_FLOW_WAIT; } } } return result; wait: { GST_DEBUG_OBJECT (jitterbuffer, "no buffer, going to wait"); if (priv->eos) { return GST_FLOW_EOS; } else { return GST_FLOW_WAIT; } } } static GstClockTime get_rtx_retry_timeout (GstRtpJitterBufferPrivate * priv) { GstClockTime rtx_retry_timeout; GstClockTime rtx_min_retry_timeout; if (priv->rtx_retry_timeout == -1) { if (priv->avg_rtx_rtt == 0) rtx_retry_timeout = DEFAULT_AUTO_RTX_TIMEOUT; else /* we want to ask for a retransmission after we waited for a * complete RTT and the additional jitter */ rtx_retry_timeout = priv->avg_rtx_rtt + priv->avg_jitter * 2; } else { rtx_retry_timeout = priv->rtx_retry_timeout * GST_MSECOND; } /* make sure we don't retry too often. On very low latency networks, * the RTT and jitter can be very low. */ if (priv->rtx_min_retry_timeout == -1) { rtx_min_retry_timeout = priv->packet_spacing; } else { rtx_min_retry_timeout = priv->rtx_min_retry_timeout * GST_MSECOND; } rtx_retry_timeout = MAX (rtx_retry_timeout, rtx_min_retry_timeout); return rtx_retry_timeout; } static GstClockTime get_rtx_retry_period (GstRtpJitterBufferPrivate * priv, GstClockTime rtx_retry_timeout) { GstClockTime rtx_retry_period; if (priv->rtx_retry_period == -1) { /* we retry up to the configured jitterbuffer size but leaving some * room for the retransmission to arrive in time */ if (rtx_retry_timeout > priv->latency_ns) { rtx_retry_period = 0; } else { rtx_retry_period = priv->latency_ns - rtx_retry_timeout; } } else { rtx_retry_period = priv->rtx_retry_period * GST_MSECOND; } return rtx_retry_period; } /* 1. For *larger* rtx-rtt, weigh a new measurement as before (1/8th) 2. For *smaller* rtx-rtt, be a bit more conservative and weigh a bit less (1/16th) 3. For very large measurements (> avg * 2), consider them "outliers" and count them a lot less (1/48th) */ static void update_avg_rtx_rtt (GstRtpJitterBufferPrivate * priv, GstClockTime rtt) { gint weight; if (priv->avg_rtx_rtt == 0) { priv->avg_rtx_rtt = rtt; return; } if (rtt > 2 * priv->avg_rtx_rtt) weight = 48; else if (rtt > priv->avg_rtx_rtt) weight = 8; else weight = 16; priv->avg_rtx_rtt = (rtt + (weight - 1) * priv->avg_rtx_rtt) / weight; } static void update_rtx_stats (GstRtpJitterBuffer * jitterbuffer, const RtpTimer * timer, GstClockTime dts, gboolean success) { GstRtpJitterBufferPrivate *priv = jitterbuffer->priv; GstClockTime delay; if (success) { /* we scheduled a retry for this packet and now we have it */ priv->num_rtx_success++; /* all the previous retry attempts failed */ priv->num_rtx_failed += timer->num_rtx_retry - 1; } else { /* All retries failed or was too late */ priv->num_rtx_failed += timer->num_rtx_retry; } /* number of retries before (hopefully) receiving the packet */ if (priv->avg_rtx_num == 0.0) priv->avg_rtx_num = timer->num_rtx_retry; else priv->avg_rtx_num = (timer->num_rtx_retry + 7 * priv->avg_rtx_num) / 8; /* Calculate the delay between retransmission request and receiving this * packet. We have a valid delay if and only if this packet is a response to * our last request. If not we don't know if this is a response to an * earlier request and delay could be way off. For RTT is more important * with correct values than to update for every packet. */ if (timer->num_rtx_retry == timer->num_rtx_received && dts != GST_CLOCK_TIME_NONE && dts > timer->rtx_last) { delay = dts - timer->rtx_last; update_avg_rtx_rtt (priv, delay); } else { delay = 0; } GST_LOG_OBJECT (jitterbuffer, "RTX #%d, result %d, success %" G_GUINT64_FORMAT ", failed %" G_GUINT64_FORMAT ", requests %" G_GUINT64_FORMAT ", dups %" G_GUINT64_FORMAT ", avg-num %g, delay %" GST_TIME_FORMAT ", avg-rtt %" GST_TIME_FORMAT, timer->seqnum, success, priv->num_rtx_success, priv->num_rtx_failed, priv->num_rtx_requests, priv->num_duplicates, priv->avg_rtx_num, GST_TIME_ARGS (delay), GST_TIME_ARGS (priv->avg_rtx_rtt)); } /* the timeout for when we expected a packet expired */ static gboolean do_expected_timeout (GstRtpJitterBuffer * jitterbuffer, RtpTimer * timer, GstClockTime now) { GstRtpJitterBufferPrivate *priv = jitterbuffer->priv; GstEvent *event; guint delay, delay_ms, avg_rtx_rtt_ms; guint rtx_retry_timeout_ms, rtx_retry_period_ms; guint rtx_deadline_ms; GstClockTime rtx_retry_period; GstClockTime rtx_retry_timeout; GstClock *clock; GstClockTimeDiff offset = 0; GST_DEBUG_OBJECT (jitterbuffer, "expected %d didn't arrive, now %" GST_TIME_FORMAT, timer->seqnum, GST_TIME_ARGS (now)); rtx_retry_timeout = get_rtx_retry_timeout (priv); rtx_retry_period = get_rtx_retry_period (priv, rtx_retry_timeout); delay = timer->rtx_delay + timer->rtx_retry; delay_ms = GST_TIME_AS_MSECONDS (delay); rtx_retry_timeout_ms = GST_TIME_AS_MSECONDS (rtx_retry_timeout); rtx_retry_period_ms = GST_TIME_AS_MSECONDS (rtx_retry_period); avg_rtx_rtt_ms = GST_TIME_AS_MSECONDS (priv->avg_rtx_rtt); rtx_deadline_ms = priv->rtx_deadline_ms != -1 ? priv->rtx_deadline_ms : priv->latency_ms; event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, gst_structure_new ("GstRTPRetransmissionRequest", "seqnum", G_TYPE_UINT, (guint) timer->seqnum, "running-time", G_TYPE_UINT64, timer->rtx_base, "delay", G_TYPE_UINT, delay_ms, "retry", G_TYPE_UINT, timer->num_rtx_retry, "frequency", G_TYPE_UINT, rtx_retry_timeout_ms, "period", G_TYPE_UINT, rtx_retry_period_ms, "deadline", G_TYPE_UINT, rtx_deadline_ms, "packet-spacing", G_TYPE_UINT64, priv->packet_spacing, "avg-rtt", G_TYPE_UINT, avg_rtx_rtt_ms, NULL)); GST_DEBUG_OBJECT (jitterbuffer, "Request RTX: %" GST_PTR_FORMAT, event); priv->num_rtx_requests++; timer->num_rtx_retry++; GST_OBJECT_LOCK (jitterbuffer); if ((clock = GST_ELEMENT_CLOCK (jitterbuffer))) { timer->rtx_last = gst_clock_get_time (clock); timer->rtx_last -= GST_ELEMENT_CAST (jitterbuffer)->base_time; } else { timer->rtx_last = now; } GST_OBJECT_UNLOCK (jitterbuffer); /* calculate the timeout for the next retransmission attempt */ timer->rtx_retry += rtx_retry_timeout; GST_DEBUG_OBJECT (jitterbuffer, "timer #%i base %" GST_TIME_FORMAT ", delay %" GST_TIME_FORMAT ", retry %" GST_TIME_FORMAT ", num_retry %u", timer->seqnum, GST_TIME_ARGS (timer->rtx_base), GST_TIME_ARGS (timer->rtx_delay), GST_TIME_ARGS (timer->rtx_retry), timer->num_rtx_retry); if ((priv->rtx_max_retries != -1 && timer->num_rtx_retry >= priv->rtx_max_retries) || (timer->rtx_retry + timer->rtx_delay > rtx_retry_period) || (timer->rtx_base + rtx_retry_period < now)) { GST_DEBUG_OBJECT (jitterbuffer, "reschedule #%i as LOST timer", timer->seqnum); /* too many retransmission request, we now convert the timer * to a lost timer, leave the num_rtx_retry as it is for stats */ timer->type = RTP_TIMER_LOST; timer->rtx_delay = 0; timer->rtx_retry = 0; offset = timeout_offset (jitterbuffer); } rtp_timer_queue_update_timer (priv->timers, timer, timer->seqnum, timer->rtx_base + timer->rtx_retry, timer->rtx_delay, offset, FALSE); JBUF_UNLOCK (priv); gst_pad_push_event (priv->sinkpad, event); JBUF_LOCK (priv); return FALSE; } /* a packet is lost */ static gboolean do_lost_timeout (GstRtpJitterBuffer * jitterbuffer, RtpTimer * timer, GstClockTime now) { GstRtpJitterBufferPrivate *priv = jitterbuffer->priv; GstClockTime timestamp; timestamp = apply_offset (jitterbuffer, get_pts_timeout (timer)); insert_lost_event (jitterbuffer, timer->seqnum, 1, timestamp, timer->duration, timer->num_rtx_retry); if (GST_CLOCK_TIME_IS_VALID (timer->rtx_last)) { /* Store info to update stats if the packet arrives too late */ timer->timeout = now + priv->rtx_stats_timeout * GST_MSECOND; timer->type = RTP_TIMER_LOST; rtp_timer_queue_insert (priv->rtx_stats_timers, timer); } else { rtp_timer_free (timer); } return TRUE; } static gboolean do_eos_timeout (GstRtpJitterBuffer * jitterbuffer, RtpTimer * timer, GstClockTime now) { GstRtpJitterBufferPrivate *priv = jitterbuffer->priv; GST_INFO_OBJECT (jitterbuffer, "got the NPT timeout"); rtp_timer_free (timer); if (!priv->eos) { GstEvent *event; /* there was no EOS in the buffer, put one in there now */ event = gst_event_new_eos (); if (priv->segment_seqnum != GST_SEQNUM_INVALID) gst_event_set_seqnum (event, priv->segment_seqnum); queue_event (jitterbuffer, event); } JBUF_SIGNAL_EVENT (priv); return TRUE; } static gboolean do_deadline_timeout (GstRtpJitterBuffer * jitterbuffer, RtpTimer * timer, GstClockTime now) { GstRtpJitterBufferPrivate *priv = jitterbuffer->priv; GST_INFO_OBJECT (jitterbuffer, "got deadline timeout"); /* timer seqnum might have been obsoleted by caps seqnum-base, * only mess with current ongoing seqnum if still unknown */ if (priv->next_seqnum == -1) priv->next_seqnum = timer->seqnum; rtp_timer_free (timer); JBUF_SIGNAL_EVENT (priv); return TRUE; } static gboolean do_timeout (GstRtpJitterBuffer * jitterbuffer, RtpTimer * timer, GstClockTime now) { gboolean removed = FALSE; switch (timer->type) { case RTP_TIMER_EXPECTED: removed = do_expected_timeout (jitterbuffer, timer, now); break; case RTP_TIMER_LOST: removed = do_lost_timeout (jitterbuffer, timer, now); break; case RTP_TIMER_DEADLINE: removed = do_deadline_timeout (jitterbuffer, timer, now); break; case RTP_TIMER_EOS: removed = do_eos_timeout (jitterbuffer, timer, now); break; } return removed; } /* called when we need to wait for the next timeout. * * We loop over the array of recorded timeouts and wait for the earliest one. * When it timed out, do the logic associated with the timer. * * If there are no timers, we wait on a gcond until something new happens. */ static void wait_next_timeout (GstRtpJitterBuffer * jitterbuffer) { GstRtpJitterBufferPrivate *priv = jitterbuffer->priv; GstClockTime now = 0; JBUF_LOCK (priv); while (priv->timer_running) { RtpTimer *timer = NULL; GQueue timers = G_QUEUE_INIT; /* don't produce data in paused */ while (priv->blocked) { JBUF_WAIT_TIMER (priv); if (!priv->timer_running) goto stopping; } /* If we have a clock, update "now" now with the very * latest running time we have. If timers are unscheduled below we * otherwise wouldn't update now (it's only updated when timers * expire), and also for the very first loop iteration now would * otherwise always be 0 */ GST_OBJECT_LOCK (jitterbuffer); if (priv->eos) { now = GST_CLOCK_TIME_NONE; } else if (GST_ELEMENT_CLOCK (jitterbuffer)) { now = gst_clock_get_time (GST_ELEMENT_CLOCK (jitterbuffer)) - GST_ELEMENT_CAST (jitterbuffer)->base_time; } GST_OBJECT_UNLOCK (jitterbuffer); GST_DEBUG_OBJECT (jitterbuffer, "now %" GST_TIME_FORMAT, GST_TIME_ARGS (now)); /* Clear expired rtx-stats timers */ if (priv->do_retransmission) rtp_timer_queue_remove_until (priv->rtx_stats_timers, now); /* Iterate expired "normal" timers */ while ((timer = rtp_timer_queue_pop_until (priv->timers, now))) { do { if (timer->type == RTP_TIMER_LOST) { GST_DEBUG_OBJECT (jitterbuffer, "Weeding out expired lost timers"); do_lost_timeout (jitterbuffer, timer, now); } else { g_queue_push_tail_link (&timers, (GList *) timer); } } while ((timer = rtp_timer_queue_pop_until (priv->timers, now))); /* execute the remaining timers */ while ((timer = (RtpTimer *) g_queue_pop_head_link (&timers))) do_timeout (jitterbuffer, timer, now); /* do_expected_timeout(), called by do_timeout will drop the * JBUF_LOCK, so we need to check if we are still running */ if (!priv->timer_running) goto stopping; } timer = rtp_timer_queue_peek_earliest (priv->timers); if (timer) { GstClock *clock; GstClockTime sync_time; GstClockID id; GstClockReturn ret; GstClockTimeDiff clock_jitter; /* we poped all immediate and due timer, so this should just never * happens */ g_assert (GST_CLOCK_TIME_IS_VALID (timer->timeout)); 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, timeout right away"); now = timer->timeout; continue; } /* prepare for sync against clock */ sync_time = timer->timeout + GST_ELEMENT_CAST (jitterbuffer)->base_time; /* add latency of peer to get input time */ sync_time += priv->peer_latency; GST_DEBUG_OBJECT (jitterbuffer, "timer #%i sync to timestamp %" GST_TIME_FORMAT " with sync time %" GST_TIME_FORMAT, timer->seqnum, GST_TIME_ARGS (get_pts_timeout (timer)), GST_TIME_ARGS (sync_time)); /* create an entry for the clock */ id = priv->clock_id = gst_clock_new_single_shot_id (clock, sync_time); priv->timer_timeout = timer->timeout; priv->timer_seqnum = timer->seqnum; 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, &clock_jitter); JBUF_LOCK (priv); if (!priv->timer_running) { gst_clock_id_unref (id); priv->clock_id = NULL; break; } if (ret != GST_CLOCK_UNSCHEDULED) { now = priv->timer_timeout + MAX (clock_jitter, 0); GST_DEBUG_OBJECT (jitterbuffer, "sync done, %d, #%d, %" GST_STIME_FORMAT, ret, priv->timer_seqnum, GST_STIME_ARGS (clock_jitter)); } else { GST_DEBUG_OBJECT (jitterbuffer, "sync unscheduled"); } /* and free the entry */ gst_clock_id_unref (id); priv->clock_id = NULL; } else { /* when draining the timers, the pusher thread will reuse our * condition to wait for completion. Signal that thread before * sleeping again here */ if (priv->eos) JBUF_SIGNAL_TIMER (priv); /* no timers, wait for activity */ JBUF_WAIT_TIMER (priv); } } stopping: JBUF_UNLOCK (priv); GST_DEBUG_OBJECT (jitterbuffer, "we are stopping"); return; } /* * This function implements the main pushing loop on the source pad. * * It first tries to push as many buffers as possible. If there is a seqnum * mismatch, we wait for the next timeouts. */ static void gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer) { GstRtpJitterBufferPrivate *priv; GstFlowReturn result = GST_FLOW_OK; priv = jitterbuffer->priv; JBUF_LOCK_CHECK (priv, flushing); do { result = handle_next_buffer (jitterbuffer); if (G_LIKELY (result == GST_FLOW_WAIT)) { /* now wait for the next event */ JBUF_SIGNAL_QUEUE (priv); JBUF_WAIT_EVENT (priv, flushing); result = GST_FLOW_OK; } } while (result == GST_FLOW_OK); /* store result for upstream */ priv->srcresult = result; /* if we get here we need to pause */ goto pause; /* ERRORS */ flushing: { result = priv->srcresult; goto pause; } pause: { GstEvent *event; JBUF_SIGNAL_QUERY (priv, FALSE); JBUF_UNLOCK (priv); GST_DEBUG_OBJECT (jitterbuffer, "pausing task, reason %s", gst_flow_get_name (result)); gst_pad_pause_task (priv->srcpad); if (result == GST_FLOW_EOS) { event = gst_event_new_eos (); if (priv->segment_seqnum != GST_SEQNUM_INVALID) gst_event_set_seqnum (event, priv->segment_seqnum); gst_pad_push_event (priv->srcpad, event); } return; } } /* collect the info from the latest RTCP packet and the jitterbuffer sync, do * some sanity checks and then emit the handle-sync signal with the parameters. * This function must be called with the LOCK */ static void do_handle_sync (GstRtpJitterBuffer * jitterbuffer) { GstRtpJitterBufferPrivate *priv; guint64 base_rtptime, base_time; guint32 clock_rate; guint64 last_rtptime; guint64 clock_base; guint64 ext_rtptime, diff; gboolean valid = TRUE, keep = FALSE; priv = jitterbuffer->priv; /* get the last values from the jitterbuffer */ rtp_jitter_buffer_get_sync (priv->jbuf, &base_rtptime, &base_time, &clock_rate, &last_rtptime); clock_base = priv->clock_base; ext_rtptime = priv->ext_rtptime; GST_DEBUG_OBJECT (jitterbuffer, "ext SR %" G_GUINT64_FORMAT ", base %" G_GUINT64_FORMAT ", clock-rate %" G_GUINT32_FORMAT ", clock-base %" G_GUINT64_FORMAT ", last-rtptime %" G_GUINT64_FORMAT, ext_rtptime, base_rtptime, clock_rate, clock_base, last_rtptime); if (base_rtptime == -1 || clock_rate == -1 || base_time == -1) { /* we keep this SR packet for later. When we get a valid RTP packet the * above values will be set and we can try to use the SR packet */ GST_DEBUG_OBJECT (jitterbuffer, "keeping for later, no RTP values"); keep = 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"); valid = FALSE; } 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 (jitterbuffer->priv->max_rtcp_rtp_time_diff != -1 && diff > gst_util_uint64_scale (jitterbuffer->priv->max_rtcp_rtp_time_diff, clock_rate, 1000)) { GST_DEBUG_OBJECT (jitterbuffer, "too far ahead"); /* should drop this, but some RTSP servers end up with bogus * way too ahead RTCP packet when repeated PAUSE/PLAY, * so still trigger rptbin sync but invalidate RTCP data * (sync might use other methods) */ ext_rtptime = -1; } GST_DEBUG_OBJECT (jitterbuffer, "ext last %" G_GUINT64_FORMAT ", diff %" G_GUINT64_FORMAT, last_rtptime, diff); } } } if (keep) { GST_DEBUG_OBJECT (jitterbuffer, "keeping RTCP packet for later"); } else if (valid) { 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, "clock-base", G_TYPE_UINT64, clock_base, "sr-ext-rtptime", G_TYPE_UINT64, ext_rtptime, "sr-buffer", GST_TYPE_BUFFER, priv->last_sr, NULL); GST_DEBUG_OBJECT (jitterbuffer, "signaling sync"); gst_buffer_replace (&priv->last_sr, NULL); JBUF_UNLOCK (priv); g_signal_emit (jitterbuffer, gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC], 0, s); JBUF_LOCK (priv); gst_structure_free (s); } else { GST_DEBUG_OBJECT (jitterbuffer, "dropping RTCP packet"); gst_buffer_replace (&priv->last_sr, NULL); } } static GstFlowReturn gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad, GstObject * parent, GstBuffer * buffer) { GstRtpJitterBuffer *jitterbuffer; GstRtpJitterBufferPrivate *priv; GstFlowReturn ret = GST_FLOW_OK; guint32 ssrc; GstRTCPPacket packet; guint64 ext_rtptime; guint32 rtptime; GstRTCPBuffer rtcp = { NULL, }; jitterbuffer = GST_RTP_JITTER_BUFFER (parent); if (G_UNLIKELY (!gst_rtcp_buffer_validate_reduced (buffer))) goto invalid_buffer; priv = jitterbuffer->priv; gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp); if (!gst_rtcp_buffer_get_first_packet (&rtcp, &packet)) goto empty_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_rtcp_buffer_unmap (&rtcp); 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); priv->ext_rtptime = ext_rtptime; gst_buffer_replace (&priv->last_sr, buffer); do_handle_sync (jitterbuffer); JBUF_UNLOCK (priv); done: gst_buffer_unref (buffer); 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; } empty_buffer: { /* this is not fatal but should be filtered earlier */ GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL), ("Received empty RTCP payload, dropping")); gst_rtcp_buffer_unmap (&rtcp); ret = GST_FLOW_OK; goto done; } ignore_buffer: { GST_DEBUG_OBJECT (jitterbuffer, "ignoring RTCP packet"); gst_rtcp_buffer_unmap (&rtcp); ret = GST_FLOW_OK; goto done; } } static gboolean gst_rtp_jitter_buffer_sink_query (GstPad * pad, GstObject * parent, GstQuery * query) { gboolean res = FALSE; GstRtpJitterBuffer *jitterbuffer; GstRtpJitterBufferPrivate *priv; jitterbuffer = GST_RTP_JITTER_BUFFER (parent); priv = jitterbuffer->priv; switch (GST_QUERY_TYPE (query)) { case GST_QUERY_CAPS: { GstCaps *filter, *caps; gst_query_parse_caps (query, &filter); caps = gst_rtp_jitter_buffer_getcaps (pad, filter); gst_query_set_caps_result (query, caps); gst_caps_unref (caps); res = TRUE; break; } default: if (GST_QUERY_IS_SERIALIZED (query)) { JBUF_LOCK_CHECK (priv, out_flushing); if (rtp_jitter_buffer_get_mode (priv->jbuf) != RTP_JITTER_BUFFER_MODE_BUFFER) { GST_DEBUG_OBJECT (jitterbuffer, "adding serialized query"); if (rtp_jitter_buffer_append_query (priv->jbuf, query)) JBUF_SIGNAL_EVENT (priv); JBUF_WAIT_QUERY (priv, out_flushing); res = priv->last_query; } else { GST_DEBUG_OBJECT (jitterbuffer, "refusing query, we are buffering"); res = FALSE; } JBUF_UNLOCK (priv); } else { res = gst_pad_query_default (pad, parent, query); } break; } return res; /* ERRORS */ out_flushing: { GST_DEBUG_OBJECT (jitterbuffer, "we are flushing"); JBUF_UNLOCK (priv); return FALSE; } } static gboolean gst_rtp_jitter_buffer_src_query (GstPad * pad, GstObject * parent, GstQuery * query) { GstRtpJitterBuffer *jitterbuffer; GstRtpJitterBufferPrivate *priv; gboolean res = FALSE; jitterbuffer = GST_RTP_JITTER_BUFFER (parent); 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, parent, 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, parent, query); } break; } case GST_QUERY_CAPS: { GstCaps *filter, *caps; gst_query_parse_caps (query, &filter); caps = gst_rtp_jitter_buffer_getcaps (pad, filter); gst_query_set_caps_result (query, caps); gst_caps_unref (caps); res = TRUE; break; } default: res = gst_pad_query_default (pad, parent, query); break; } 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); if (priv->max_ts_offset_adjustment != 0) { gint64 new_offset = g_value_get_int64 (value); if (new_offset > priv->ts_offset) { priv->ts_offset_remainder = new_offset - priv->ts_offset; } else { priv->ts_offset_remainder = -(priv->ts_offset - new_offset); } } else { priv->ts_offset = g_value_get_int64 (value); priv->ts_offset_remainder = 0; update_timer_offsets (jitterbuffer); } priv->ts_discont = TRUE; JBUF_UNLOCK (priv); break; case PROP_MAX_TS_OFFSET_ADJUSTMENT: JBUF_LOCK (priv); priv->max_ts_offset_adjustment = g_value_get_uint64 (value); 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_POST_DROP_MESSAGES: JBUF_LOCK (priv); priv->post_drop_messages = g_value_get_boolean (value); JBUF_UNLOCK (priv); break; case PROP_DROP_MESSAGES_INTERVAL: JBUF_LOCK (priv); priv->drop_messages_interval_ms = g_value_get_uint (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; case PROP_DO_RETRANSMISSION: JBUF_LOCK (priv); priv->do_retransmission = g_value_get_boolean (value); JBUF_UNLOCK (priv); break; case PROP_RTX_NEXT_SEQNUM: JBUF_LOCK (priv); priv->rtx_next_seqnum = g_value_get_boolean (value); JBUF_UNLOCK (priv); break; case PROP_RTX_DELAY: JBUF_LOCK (priv); priv->rtx_delay = g_value_get_int (value); JBUF_UNLOCK (priv); break; case PROP_RTX_MIN_DELAY: JBUF_LOCK (priv); priv->rtx_min_delay = g_value_get_uint (value); JBUF_UNLOCK (priv); break; case PROP_RTX_DELAY_REORDER: JBUF_LOCK (priv); priv->rtx_delay_reorder = g_value_get_int (value); JBUF_UNLOCK (priv); break; case PROP_RTX_RETRY_TIMEOUT: JBUF_LOCK (priv); priv->rtx_retry_timeout = g_value_get_int (value); JBUF_UNLOCK (priv); break; case PROP_RTX_MIN_RETRY_TIMEOUT: JBUF_LOCK (priv); priv->rtx_min_retry_timeout = g_value_get_int (value); JBUF_UNLOCK (priv); break; case PROP_RTX_RETRY_PERIOD: JBUF_LOCK (priv); priv->rtx_retry_period = g_value_get_int (value); JBUF_UNLOCK (priv); break; case PROP_RTX_MAX_RETRIES: JBUF_LOCK (priv); priv->rtx_max_retries = g_value_get_int (value); JBUF_UNLOCK (priv); break; case PROP_RTX_DEADLINE: JBUF_LOCK (priv); priv->rtx_deadline_ms = g_value_get_int (value); JBUF_UNLOCK (priv); break; case PROP_RTX_STATS_TIMEOUT: JBUF_LOCK (priv); priv->rtx_stats_timeout = g_value_get_uint (value); JBUF_UNLOCK (priv); break; case PROP_MAX_RTCP_RTP_TIME_DIFF: JBUF_LOCK (priv); priv->max_rtcp_rtp_time_diff = g_value_get_int (value); JBUF_UNLOCK (priv); break; case PROP_MAX_DROPOUT_TIME: JBUF_LOCK (priv); priv->max_dropout_time = g_value_get_uint (value); JBUF_UNLOCK (priv); break; case PROP_MAX_MISORDER_TIME: JBUF_LOCK (priv); priv->max_misorder_time = g_value_get_uint (value); JBUF_UNLOCK (priv); break; case PROP_RFC7273_SYNC: JBUF_LOCK (priv); rtp_jitter_buffer_set_rfc7273_sync (priv->jbuf, g_value_get_boolean (value)); JBUF_UNLOCK (priv); break; case PROP_FASTSTART_MIN_PACKETS: JBUF_LOCK (priv); priv->faststart_min_packets = g_value_get_uint (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_MAX_TS_OFFSET_ADJUSTMENT: JBUF_LOCK (priv); g_value_set_uint64 (value, priv->max_ts_offset_adjustment); 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_POST_DROP_MESSAGES: JBUF_LOCK (priv); g_value_set_boolean (value, priv->post_drop_messages); JBUF_UNLOCK (priv); break; case PROP_DROP_MESSAGES_INTERVAL: JBUF_LOCK (priv); g_value_set_uint (value, priv->drop_messages_interval_ms); 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; } case PROP_DO_RETRANSMISSION: JBUF_LOCK (priv); g_value_set_boolean (value, priv->do_retransmission); JBUF_UNLOCK (priv); break; case PROP_RTX_NEXT_SEQNUM: JBUF_LOCK (priv); g_value_set_boolean (value, priv->rtx_next_seqnum); JBUF_UNLOCK (priv); break; case PROP_RTX_DELAY: JBUF_LOCK (priv); g_value_set_int (value, priv->rtx_delay); JBUF_UNLOCK (priv); break; case PROP_RTX_MIN_DELAY: JBUF_LOCK (priv); g_value_set_uint (value, priv->rtx_min_delay); JBUF_UNLOCK (priv); break; case PROP_RTX_DELAY_REORDER: JBUF_LOCK (priv); g_value_set_int (value, priv->rtx_delay_reorder); JBUF_UNLOCK (priv); break; case PROP_RTX_RETRY_TIMEOUT: JBUF_LOCK (priv); g_value_set_int (value, priv->rtx_retry_timeout); JBUF_UNLOCK (priv); break; case PROP_RTX_MIN_RETRY_TIMEOUT: JBUF_LOCK (priv); g_value_set_int (value, priv->rtx_min_retry_timeout); JBUF_UNLOCK (priv); break; case PROP_RTX_RETRY_PERIOD: JBUF_LOCK (priv); g_value_set_int (value, priv->rtx_retry_period); JBUF_UNLOCK (priv); break; case PROP_RTX_MAX_RETRIES: JBUF_LOCK (priv); g_value_set_int (value, priv->rtx_max_retries); JBUF_UNLOCK (priv); break; case PROP_RTX_DEADLINE: JBUF_LOCK (priv); g_value_set_int (value, priv->rtx_deadline_ms); JBUF_UNLOCK (priv); break; case PROP_RTX_STATS_TIMEOUT: JBUF_LOCK (priv); g_value_set_uint (value, priv->rtx_stats_timeout); JBUF_UNLOCK (priv); break; case PROP_STATS: g_value_take_boxed (value, gst_rtp_jitter_buffer_create_stats (jitterbuffer)); break; case PROP_MAX_RTCP_RTP_TIME_DIFF: JBUF_LOCK (priv); g_value_set_int (value, priv->max_rtcp_rtp_time_diff); JBUF_UNLOCK (priv); break; case PROP_MAX_DROPOUT_TIME: JBUF_LOCK (priv); g_value_set_uint (value, priv->max_dropout_time); JBUF_UNLOCK (priv); break; case PROP_MAX_MISORDER_TIME: JBUF_LOCK (priv); g_value_set_uint (value, priv->max_misorder_time); JBUF_UNLOCK (priv); break; case PROP_RFC7273_SYNC: JBUF_LOCK (priv); g_value_set_boolean (value, rtp_jitter_buffer_get_rfc7273_sync (priv->jbuf)); JBUF_UNLOCK (priv); break; case PROP_FASTSTART_MIN_PACKETS: JBUF_LOCK (priv); g_value_set_uint (value, priv->faststart_min_packets); JBUF_UNLOCK (priv); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static GstStructure * gst_rtp_jitter_buffer_create_stats (GstRtpJitterBuffer * jbuf) { GstRtpJitterBufferPrivate *priv = jbuf->priv; GstStructure *s; JBUF_LOCK (priv); s = gst_structure_new ("application/x-rtp-jitterbuffer-stats", "num-pushed", G_TYPE_UINT64, priv->num_pushed, "num-lost", G_TYPE_UINT64, priv->num_lost, "num-late", G_TYPE_UINT64, priv->num_late, "num-duplicates", G_TYPE_UINT64, priv->num_duplicates, "avg-jitter", G_TYPE_UINT64, priv->avg_jitter, "rtx-count", G_TYPE_UINT64, priv->num_rtx_requests, "rtx-success-count", G_TYPE_UINT64, priv->num_rtx_success, "rtx-per-packet", G_TYPE_DOUBLE, priv->avg_rtx_num, "rtx-rtt", G_TYPE_UINT64, priv->avg_rtx_rtt, NULL); JBUF_UNLOCK (priv); return s; }