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gstreamer/gst/rtpmanager/gstrtpjitterbuffer.c
Wim Taymans eee515cb2c rtpjitterbuffer: serialize events in the buffer 
Serialize events into the jitterbuffer by inserting them with a -1
seqnum.
Update unit test to expect events from the streaming thread.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=652986
2013-12-10 11:57:37 +01:00

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/*
* Farsight Voice+Video library
*
* Copyright 2007 Collabora Ltd,
* Copyright 2007 Nokia Corporation
* @author: Philippe Kalaf <philippe.kalaf@collabora.co.uk>.
* Copyright 2007 Wim Taymans <wim.taymans@gmail.com>
*
* 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
*
* 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 incomming
* 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 abonormally 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.
*
* <refsect2>
* <title>Example pipelines</title>
* |[
* 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.
* </refsect2>
*
* Last reviewed on 2007-05-28 (0.10.5)
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <stdlib.h>
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include "gstrtpjitterbuffer.h"
#include "rtpjitterbuffer.h"
#include "rtpstats.h"
#include <gst/glib-compat-private.h>
GST_DEBUG_CATEGORY (rtpjitterbuffer_debug);
#define GST_CAT_DEFAULT (rtpjitterbuffer_debug)
/* RTPJitterBuffer signals and args */
enum
{
SIGNAL_REQUEST_PT_MAP,
SIGNAL_CLEAR_PT_MAP,
SIGNAL_HANDLE_SYNC,
SIGNAL_ON_NPT_STOP,
SIGNAL_SET_ACTIVE,
LAST_SIGNAL
};
#define DEFAULT_LATENCY_MS 200
#define DEFAULT_DROP_ON_LATENCY FALSE
#define DEFAULT_TS_OFFSET 0
#define DEFAULT_DO_LOST FALSE
#define DEFAULT_MODE RTP_JITTER_BUFFER_MODE_SLAVE
#define DEFAULT_PERCENT 0
#define DEFAULT_DO_RETRANSMISSION FALSE
#define DEFAULT_RTX_DELAY 20
#define DEFAULT_RTX_DELAY_REORDER 3
#define DEFAULT_RTX_RETRY_TIMEOUT 40
#define DEFAULT_RTX_RETRY_PERIOD 160
enum
{
PROP_0,
PROP_LATENCY,
PROP_DROP_ON_LATENCY,
PROP_TS_OFFSET,
PROP_DO_LOST,
PROP_MODE,
PROP_PERCENT,
PROP_DO_RETRANSMISSION,
PROP_RTX_DELAY,
PROP_RTX_DELAY_REORDER,
PROP_RTX_RETRY_TIMEOUT,
PROP_RTX_RETRY_PERIOD,
PROP_STATS,
PROP_LAST
};
#define JBUF_LOCK(priv) (g_mutex_lock (&(priv)->jbuf_lock))
#define JBUF_LOCK_CHECK(priv,label) G_STMT_START { \
JBUF_LOCK (priv); \
if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
goto label; \
} G_STMT_END
#define JBUF_UNLOCK(priv) (g_mutex_unlock (&(priv)->jbuf_lock))
#define JBUF_WAIT_TIMER(priv) G_STMT_START { \
GST_DEBUG ("waiting timer"); \
(priv)->waiting_timer = TRUE; \
g_cond_wait (&(priv)->jbuf_timer, &(priv)->jbuf_lock); \
(priv)->waiting_timer = FALSE; \
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"); \
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
struct _GstRtpJitterBufferPrivate
{
GstPad *sinkpad, *srcpad;
GstPad *rtcpsinkpad;
RTPJitterBuffer *jbuf;
GMutex jbuf_lock;
gboolean waiting_timer;
GCond jbuf_timer;
gboolean waiting_event;
GCond jbuf_event;
gboolean discont;
gboolean ts_discont;
gboolean active;
guint64 out_offset;
gboolean timer_running;
GThread *timer_thread;
/* properties */
guint latency_ms;
guint64 latency_ns;
gboolean drop_on_latency;
gint64 ts_offset;
gboolean do_lost;
gboolean do_retransmission;
gint rtx_delay;
gint rtx_delay_reorder;
gint rtx_retry_timeout;
gint rtx_retry_period;
/* the last seqnum we pushed out */
guint32 last_popped_seqnum;
/* the next expected seqnum we push */
guint32 next_seqnum;
/* last output time */
GstClockTime last_out_time;
/* last valid input timestamp and rtptime pair */
GstClockTime ips_dts;
guint64 ips_rtptime;
GstClockTime packet_spacing;
/* the next expected seqnum we receive */
GstClockTime last_in_dts;
guint32 last_in_seqnum;
guint32 next_in_seqnum;
GArray *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;
/* clock rate and rtp timestamp offset */
gint last_pt;
gint32 clock_rate;
gint64 clock_base;
gint64 prev_ts_offset;
/* when we are shutting down */
GstFlowReturn srcresult;
gboolean blocked;
/* for sync */
GstSegment segment;
GstClockID clock_id;
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_late;
guint64 num_duplicates;
guint64 num_rtx_requests;
guint64 num_rtx_success;
guint64 num_rtx_failed;
gdouble avg_rtx_num;
guint64 avg_rtx_rtt;
};
typedef enum
{
TIMER_TYPE_EXPECTED,
TIMER_TYPE_LOST,
TIMER_TYPE_DEADLINE,
TIMER_TYPE_EOS
} TimerType;
typedef struct
{
guint idx;
guint16 seqnum;
guint num;
TimerType type;
GstClockTime timeout;
GstClockTime duration;
GstClockTime rtx_base;
GstClockTime rtx_delay;
GstClockTime rtx_retry;
GstClockTime rtx_last;
guint num_rtx_retry;
} TimerData;
#define GST_RTP_JITTER_BUFFER_GET_PRIVATE(o) \
(G_TYPE_INSTANCE_GET_PRIVATE ((o), GST_TYPE_RTP_JITTER_BUFFER, \
GstRtpJitterBufferPrivate))
static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"clock-rate = (int) [ 1, 2147483647 ]"
/* "payload = (int) , "
* "encoding-name = (string) "
*/ )
);
static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_rtcp_template =
GST_STATIC_PAD_TEMPLATE ("sink_rtcp",
GST_PAD_SINK,
GST_PAD_REQUEST,
GST_STATIC_CAPS ("application/x-rtcp")
);
static GstStaticPadTemplate gst_rtp_jitter_buffer_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp"
/* "payload = (int) , "
* "clock-rate = (int) , "
* "encoding-name = (string) "
*/ )
);
static guint gst_rtp_jitter_buffer_signals[LAST_SIGNAL] = { 0 };
#define gst_rtp_jitter_buffer_parent_class parent_class
G_DEFINE_TYPE (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);
/* 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 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 remove_all_timers (GstRtpJitterBuffer * jitterbuffer);
static void wait_next_timeout (GstRtpJitterBuffer * jitterbuffer);
static GstStructure *gst_rtp_jitter_buffer_create_stats (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;
g_type_class_add_private (klass, sizeof (GstRtpJitterBufferPrivate));
gobject_class->finalize = gst_rtp_jitter_buffer_finalize;
gobject_class->set_property = gst_rtp_jitter_buffer_set_property;
gobject_class->get_property = gst_rtp_jitter_buffer_get_property;
/**
* GstRtpJitterBuffer:latency:
*
* The maximum latency of the jitterbuffer. Packets will be kept in the buffer
* for at most this time.
*/
g_object_class_install_property (gobject_class, PROP_LATENCY,
g_param_spec_uint ("latency", "Buffer latency in ms",
"Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRtpJitterBuffer:drop-on-latency:
*
* Drop oldest buffers when the queue is completely filled.
*/
g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
g_param_spec_boolean ("drop-on-latency",
"Drop buffers when maximum latency is reached",
"Tells the jitterbuffer to never exceed the given latency in size",
DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRtpJitterBuffer:ts-offset:
*
* Adjust GStreamer output buffer timestamps in the jitterbuffer with offset.
* This is mainly used to ensure interstream synchronisation.
*/
g_object_class_install_property (gobject_class, PROP_TS_OFFSET,
g_param_spec_int64 ("ts-offset", "Timestamp Offset",
"Adjust buffer timestamps with offset in nanoseconds", G_MININT64,
G_MAXINT64, DEFAULT_TS_OFFSET,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRtpJitterBuffer:do-lost:
*
* Send out a GstRTPPacketLost event downstream when a packet is considered
* lost.
*/
g_object_class_install_property (gobject_class, PROP_DO_LOST,
g_param_spec_boolean ("do-lost", "Do Lost",
"Send an event downstream when a packet is lost", DEFAULT_DO_LOST,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRtpJitterBuffer:mode:
*
* Control the buffering and timestamping mode used by the jitterbuffer.
*/
g_object_class_install_property (gobject_class, PROP_MODE,
g_param_spec_enum ("mode", "Mode",
"Control the buffering algorithm in use", RTP_TYPE_JITTER_BUFFER_MODE,
DEFAULT_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRtpJitterBuffer:percent:
*
* The percent of the jitterbuffer that is filled.
*/
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-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-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.
*
* 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 (-1 automatic)",
-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-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:stats:
*
* Various jitterbuffer statistics. This property returns a GstStructure
* with name application/x-rtp-jitterbuffer-stats with the following fields:
*
* "rtx-count" G_TYPE_UINT64 The number of retransmissions requested
* "rtx-success-count" G_TYPE_UINT64 The number of successful retransmissions
* "rtx-per-packet" G_TYPE_DOUBLE Average number of RTX per packet
* "rtx-rtt" G_TYPE_UINT64 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::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, g_cclosure_marshal_generic,
GST_TYPE_CAPS, 1, G_TYPE_UINT);
/**
* GstRtpJitterBuffer::handle-sync:
* @buffer: the object which received the signal
* @struct: a GstStructure containing sync values.
*
* Be notified of new sync values.
*/
gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC] =
g_signal_new ("handle-sync", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
handle_sync), NULL, NULL, g_cclosure_marshal_VOID__BOXED,
G_TYPE_NONE, 1, GST_TYPE_STRUCTURE | G_SIGNAL_TYPE_STATIC_SCOPE);
/**
* GstRtpJitterBuffer::on-npt-stop:
* @buffer: the object which received the signal
*
* Signal that the jitterbufer has pushed the RTP packet that corresponds to
* the npt-stop position.
*/
gst_rtp_jitter_buffer_signals[SIGNAL_ON_NPT_STOP] =
g_signal_new ("on-npt-stop", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
on_npt_stop), NULL, NULL, g_cclosure_marshal_VOID__VOID,
G_TYPE_NONE, 0, G_TYPE_NONE);
/**
* GstRtpJitterBuffer::clear-pt-map:
* @buffer: the object which received the signal
*
* Invalidate the clock-rate as obtained with the
* #GstRtpJitterBuffer::request-pt-map signal.
*/
gst_rtp_jitter_buffer_signals[SIGNAL_CLEAR_PT_MAP] =
g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
G_STRUCT_OFFSET (GstRtpJitterBufferClass, clear_pt_map), NULL, NULL,
g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
/**
* GstRtpJitterBuffer::set-active:
* @buffer: the object which received the signal
*
* Start pushing out packets with the given base time. This signal is only
* useful in buffering mode.
*
* Returns: the time of the last pushed packet.
*/
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,
g_cclosure_marshal_generic, 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);
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_rtp_jitter_buffer_src_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_rtp_jitter_buffer_sink_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&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 <philippe.kalaf@collabora.co.uk>, "
"Wim Taymans <wim.taymans@gmail.com>");
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");
}
static void
gst_rtp_jitter_buffer_init (GstRtpJitterBuffer * jitterbuffer)
{
GstRtpJitterBufferPrivate *priv;
priv = GST_RTP_JITTER_BUFFER_GET_PRIVATE (jitterbuffer);
jitterbuffer->priv = priv;
priv->latency_ms = DEFAULT_LATENCY_MS;
priv->latency_ns = priv->latency_ms * GST_MSECOND;
priv->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
priv->do_lost = DEFAULT_DO_LOST;
priv->do_retransmission = DEFAULT_DO_RETRANSMISSION;
priv->rtx_delay = DEFAULT_RTX_DELAY;
priv->rtx_delay_reorder = DEFAULT_RTX_DELAY_REORDER;
priv->rtx_retry_timeout = DEFAULT_RTX_RETRY_TIMEOUT;
priv->rtx_retry_period = DEFAULT_RTX_RETRY_PERIOD;
priv->timers = g_array_new (FALSE, TRUE, sizeof (TimerData));
priv->jbuf = rtp_jitter_buffer_new ();
g_mutex_init (&priv->jbuf_lock);
g_cond_init (&priv->jbuf_timer);
g_cond_init (&priv->jbuf_event);
/* reset skew detection initialy */
rtp_jitter_buffer_reset_skew (priv->jbuf);
rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
rtp_jitter_buffer_set_buffering (priv->jbuf, FALSE);
priv->active = TRUE;
priv->srcpad =
gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_src_template,
"src");
gst_pad_set_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_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);
}
#define ITEM_TYPE_BUFFER 0
#define ITEM_TYPE_LOST 1
#define ITEM_TYPE_EVENT 2
static RTPJitterBufferItem *
alloc_item (gpointer data, guint type, GstClockTime dts, GstClockTime pts,
guint seqnum, guint count, guint rtptime)
{
RTPJitterBufferItem *item;
item = g_slice_new (RTPJitterBufferItem);
item->data = data;
item->next = NULL;
item->prev = NULL;
item->type = type;
item->dts = dts;
item->pts = pts;
item->seqnum = seqnum;
item->count = count;
item->rtptime = rtptime;
return item;
}
static void
free_item (RTPJitterBufferItem * item)
{
if (item->data)
gst_mini_object_unref (item->data);
g_slice_free (RTPJitterBufferItem, item);
}
static void
gst_rtp_jitter_buffer_finalize (GObject * object)
{
GstRtpJitterBuffer *jitterbuffer;
GstRtpJitterBufferPrivate *priv;
jitterbuffer = GST_RTP_JITTER_BUFFER (object);
priv = jitterbuffer->priv;
g_array_free (priv->timers, TRUE);
g_mutex_clear (&priv->jbuf_lock);
g_cond_clear (&priv->jbuf_timer);
g_cond_clear (&priv->jbuf_event);
rtp_jitter_buffer_flush (priv->jbuf, (GFunc) free_item, NULL);
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;
GValue val = { 0, };
jitterbuffer = GST_RTP_JITTER_BUFFER (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) {
otherpad = 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 (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 (element);
priv = jitterbuffer->priv;
GST_DEBUG_OBJECT (element, "releasing pad %s:%s", GST_DEBUG_PAD_NAME (pad));
if (priv->rtcpsinkpad == pad) {
remove_rtcp_sink (jitterbuffer);
} else
goto wrong_pad;
return;
/* ERRORS */
wrong_pad:
{
g_warning ("gstjitterbuffer: asked to release an unknown pad");
return;
}
}
static GstClock *
gst_rtp_jitter_buffer_provide_clock (GstElement * element)
{
return gst_system_clock_obtain ();
}
static void
gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer)
{
GstRtpJitterBufferPrivate *priv;
priv = jitterbuffer->priv;
/* this will trigger a new pt-map request signal, FIXME, do something better. */
JBUF_LOCK (priv);
priv->clock_rate = -1;
/* 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->dts + 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;
}
/*
* Must be called with JBUF_LOCK held
*/
static gboolean
gst_jitter_buffer_sink_parse_caps (GstRtpJitterBuffer * jitterbuffer,
GstCaps * caps)
{
GstRtpJitterBufferPrivate *priv;
GstStructure *caps_struct;
guint val;
GstClockTime tval;
priv = jitterbuffer->priv;
/* first parse the caps */
caps_struct = gst_caps_get_structure (caps, 0);
GST_DEBUG_OBJECT (jitterbuffer, "got caps");
/* we need a clock-rate to convert the rtp timestamps to GStreamer time and to
* measure the amount of data in the buffer */
if (!gst_structure_get_int (caps_struct, "clock-rate", &priv->clock_rate))
goto error;
if (priv->clock_rate <= 0)
goto wrong_rate;
GST_DEBUG_OBJECT (jitterbuffer, "got clock-rate %d", priv->clock_rate);
rtp_jitter_buffer_set_clock_rate (priv->jbuf, priv->clock_rate);
/* The clock base is the RTP timestamp corrsponding to the npt-start value. We
* can use this to track the amount of time elapsed on the sender. */
if (gst_structure_get_uint (caps_struct, "clock-base", &val))
priv->clock_base = val;
else
priv->clock_base = -1;
priv->ext_timestamp = priv->clock_base;
GST_DEBUG_OBJECT (jitterbuffer, "got clock-base %" G_GINT64_FORMAT,
priv->clock_base);
if (gst_structure_get_uint (caps_struct, "seqnum-base", &val)) {
/* first expected seqnum, only update when we didn't have a previous base. */
if (priv->next_in_seqnum == -1)
priv->next_in_seqnum = val;
if (priv->next_seqnum == -1)
priv->next_seqnum = val;
}
GST_DEBUG_OBJECT (jitterbuffer, "got seqnum-base %d", priv->next_in_seqnum);
/* the start and stop times. The seqnum-base corresponds to the start time. We
* will keep track of the seqnums on the output and when we reach the one
* corresponding to npt-stop, we emit the npt-stop-reached signal */
if (gst_structure_get_clock_time (caps_struct, "npt-start", &tval))
priv->npt_start = tval;
else
priv->npt_start = 0;
if (gst_structure_get_clock_time (caps_struct, "npt-stop", &tval))
priv->npt_stop = tval;
else
priv->npt_stop = -1;
GST_DEBUG_OBJECT (jitterbuffer,
"npt start/stop: %" GST_TIME_FORMAT "-%" GST_TIME_FORMAT,
GST_TIME_ARGS (priv->npt_start), GST_TIME_ARGS (priv->npt_stop));
return TRUE;
/* ERRORS */
error:
{
GST_DEBUG_OBJECT (jitterbuffer, "No clock-rate in caps!");
return FALSE;
}
wrong_rate:
{
GST_DEBUG_OBJECT (jitterbuffer, "Invalid clock-rate %d", priv->clock_rate);
return FALSE;
}
}
static 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_UNLOCK (priv);
}
static void
gst_rtp_jitter_buffer_flush_stop (GstRtpJitterBuffer * jitterbuffer)
{
GstRtpJitterBufferPrivate *priv;
priv = jitterbuffer->priv;
JBUF_LOCK (priv);
GST_DEBUG_OBJECT (jitterbuffer, "Enabling pop on queue");
/* Mark as non flushing */
priv->srcresult = GST_FLOW_OK;
gst_segment_init (&priv->segment, GST_FORMAT_TIME);
priv->last_popped_seqnum = -1;
priv->last_out_time = -1;
priv->next_seqnum = -1;
priv->ips_rtptime = -1;
priv->ips_dts = GST_CLOCK_TIME_NONE;
priv->packet_spacing = 0;
priv->next_in_seqnum = -1;
priv->clock_rate = -1;
priv->eos = FALSE;
priv->estimated_eos = -1;
priv->last_elapsed = 0;
priv->ext_timestamp = -1;
GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
rtp_jitter_buffer_flush (priv->jbuf, (GFunc) free_item, NULL);
rtp_jitter_buffer_reset_skew (priv->jbuf);
remove_all_timers (jitterbuffer);
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->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;
unschedule_current_timer (jitterbuffer);
JBUF_SIGNAL_TIMER (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 (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;
RTPJitterBufferItem *item;
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_CAPS:
{
GstCaps *caps;
gst_event_parse_caps (event, &caps);
if (!gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps))
goto wrong_caps;
break;
}
case GST_EVENT_SEGMENT:
gst_event_copy_segment (event, &priv->segment);
/* we need time for now */
if (priv->segment.format != GST_FORMAT_TIME)
goto newseg_wrong_format;
GST_DEBUG_OBJECT (jitterbuffer,
"newsegment: %" GST_SEGMENT_FORMAT, &priv->segment);
break;
case GST_EVENT_EOS:
priv->eos = TRUE;
break;
default:
break;
}
GST_DEBUG_OBJECT (jitterbuffer, "adding event");
item = alloc_item (event, ITEM_TYPE_EVENT, -1, -1, -1, 0, -1);
rtp_jitter_buffer_insert (priv->jbuf, item, NULL, NULL);
JBUF_SIGNAL_EVENT (priv);
return TRUE;
/* ERRORS */
wrong_caps:
{
GST_DEBUG_OBJECT (jitterbuffer, "received invalid caps");
gst_event_unref (event);
return FALSE;
}
newseg_wrong_format:
{
GST_DEBUG_OBJECT (jitterbuffer, "received non TIME newsegment");
gst_event_unref (event);
return FALSE;
}
}
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 emiting 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);
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 void
check_buffering_percent (GstRtpJitterBuffer * jitterbuffer, gint * percent)
{
GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
/* too short a stream, or too close to EOS will never really fill buffer */
if (*percent != -1 && priv->npt_stop != -1 &&
priv->npt_stop - priv->npt_start <=
rtp_jitter_buffer_get_delay (priv->jbuf)) {
GST_DEBUG_OBJECT (jitterbuffer, "short stream; faking full buffer");
rtp_jitter_buffer_set_buffering (priv->jbuf, FALSE);
*percent = 100;
}
}
static void
post_buffering_percent (GstRtpJitterBuffer * jitterbuffer, gint percent)
{
GstMessage *message;
/* Post a buffering message */
message = gst_message_new_buffering (GST_OBJECT_CAST (jitterbuffer), percent);
gst_message_set_buffering_stats (message, GST_BUFFERING_LIVE, -1, -1, -1);
gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), message);
}
static 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 TimerData *
find_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type, guint16 seqnum)
{
GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
TimerData *timer = NULL;
gint i, len;
len = priv->timers->len;
for (i = 0; i < len; i++) {
TimerData *test = &g_array_index (priv->timers, TimerData, i);
if (test->seqnum == seqnum && test->type == type) {
timer = test;
break;
}
}
return timer;
}
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 GstClockTime
get_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
{
GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
GstClockTime test_timeout;
if ((test_timeout = timer->timeout) == -1)
return -1;
if (timer->type != TIMER_TYPE_EXPECTED) {
/* add our latency and offset to get output times. */
test_timeout = apply_offset (jitterbuffer, test_timeout);
test_timeout += priv->latency_ns;
}
return test_timeout;
}
static void
recalculate_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
{
GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
if (priv->clock_id) {
GstClockTime timeout = get_timeout (jitterbuffer, timer);
GST_DEBUG ("%" GST_TIME_FORMAT " <> %" GST_TIME_FORMAT,
GST_TIME_ARGS (timeout), GST_TIME_ARGS (priv->timer_timeout));
if (timeout == -1 || timeout < priv->timer_timeout)
unschedule_current_timer (jitterbuffer);
}
}
static TimerData *
add_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type,
guint16 seqnum, guint num, GstClockTime timeout, GstClockTime delay,
GstClockTime duration)
{
GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
TimerData *timer;
gint len;
GST_DEBUG_OBJECT (jitterbuffer,
"add timer for seqnum %d to %" GST_TIME_FORMAT ", delay %"
GST_TIME_FORMAT, seqnum, GST_TIME_ARGS (timeout), GST_TIME_ARGS (delay));
len = priv->timers->len;
g_array_set_size (priv->timers, len + 1);
timer = &g_array_index (priv->timers, TimerData, len);
timer->idx = len;
timer->type = type;
timer->seqnum = seqnum;
timer->num = num;
timer->timeout = timeout + delay;
timer->duration = duration;
if (type == TIMER_TYPE_EXPECTED) {
timer->rtx_base = timeout;
timer->rtx_delay = delay;
timer->rtx_retry = 0;
}
timer->num_rtx_retry = 0;
recalculate_timer (jitterbuffer, timer);
JBUF_SIGNAL_TIMER (priv);
return timer;
}
static void
reschedule_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
guint16 seqnum, GstClockTime timeout, GstClockTime delay, gboolean reset)
{
GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
gboolean seqchange, timechange;
guint16 oldseq;
seqchange = timer->seqnum != seqnum;
timechange = timer->timeout != timeout;
if (!seqchange && !timechange)
return;
oldseq = timer->seqnum;
GST_DEBUG_OBJECT (jitterbuffer,
"replace timer for seqnum %d->%d to %" GST_TIME_FORMAT,
oldseq, seqnum, GST_TIME_ARGS (timeout + delay));
timer->timeout = timeout + delay;
timer->seqnum = seqnum;
if (reset) {
timer->rtx_base = timeout;
timer->rtx_delay = delay;
timer->rtx_retry = 0;
}
if (priv->clock_id) {
/* we changed the seqnum and there is a timer currently waiting with this
* seqnum, unschedule it */
if (seqchange && priv->timer_seqnum == oldseq)
unschedule_current_timer (jitterbuffer);
/* we changed the time, check if it is earlier than what we are waiting
* for and unschedule if so */
else if (timechange)
recalculate_timer (jitterbuffer, timer);
}
}
static TimerData *
set_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type,
guint16 seqnum, GstClockTime timeout)
{
TimerData *timer;
/* find the seqnum timer */
timer = find_timer (jitterbuffer, type, seqnum);
if (timer == NULL) {
timer = add_timer (jitterbuffer, type, seqnum, 0, timeout, 0, -1);
} else {
reschedule_timer (jitterbuffer, timer, seqnum, timeout, 0, FALSE);
}
return timer;
}
static void
remove_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
{
GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
guint idx;
if (priv->clock_id && priv->timer_seqnum == timer->seqnum)
unschedule_current_timer (jitterbuffer);
idx = timer->idx;
GST_DEBUG_OBJECT (jitterbuffer, "removed index %d", idx);
g_array_remove_index_fast (priv->timers, idx);
timer->idx = idx;
}
static void
remove_all_timers (GstRtpJitterBuffer * jitterbuffer)
{
GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
GST_DEBUG_OBJECT (jitterbuffer, "removed all timers");
g_array_set_size (priv->timers, 0);
unschedule_current_timer (jitterbuffer);
}
/* 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, gboolean do_next_seqnum)
{
GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
TimerData *timer = NULL;
gint i, len;
/* go through all timers and unschedule the ones with a large gap, also find
* the timer for the seqnum */
len = priv->timers->len;
for (i = 0; i < len; i++) {
TimerData *test = &g_array_index (priv->timers, TimerData, i);
gint gap;
gap = gst_rtp_buffer_compare_seqnum (test->seqnum, seqnum);
GST_DEBUG_OBJECT (jitterbuffer, "%d, #%d<->#%d gap %d", i,
test->seqnum, seqnum, gap);
if (gap == 0) {
GST_DEBUG ("found timer for current seqnum");
/* the timer for the current seqnum */
timer = test;
} else if (gap > priv->rtx_delay_reorder) {
/* 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 == TIMER_TYPE_EXPECTED)
reschedule_timer (jitterbuffer, test, test->seqnum, -1, 0, FALSE);
}
}
if (priv->packet_spacing > 0 && do_next_seqnum && priv->do_retransmission) {
GstClockTime expected, delay;
/* calculate expected arrival time of the next seqnum */
expected = dts + priv->packet_spacing;
delay = priv->rtx_delay * GST_MSECOND;
/* and update/install timer for next seqnum */
if (timer)
reschedule_timer (jitterbuffer, timer, priv->next_in_seqnum, expected,
delay, TRUE);
else
add_timer (jitterbuffer, TIMER_TYPE_EXPECTED, priv->next_in_seqnum, 0,
expected, delay, priv->packet_spacing);
} else if (timer && timer->type != TIMER_TYPE_DEADLINE) {
if (timer->num_rtx_retry > 0) {
GstClockTime rtx_last;
/* 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;
/* number of retries before 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, start with when we scheduled this timeout last */
rtx_last = timer->rtx_last;
if (dts > rtx_last) {
GstClockTime delay;
/* we have a valid delay if this packet arrived after we scheduled the
* request */
delay = dts - rtx_last;
if (priv->avg_rtx_rtt == 0)
priv->avg_rtx_rtt = delay;
else
priv->avg_rtx_rtt = (delay + 7 * priv->avg_rtx_rtt) / 8;
}
GST_LOG_OBJECT (jitterbuffer,
"RTX success %" G_GUINT64_FORMAT ", failed %" G_GUINT64_FORMAT
", requests %" G_GUINT64_FORMAT ", dups %" G_GUINT64_FORMAT
", avg-num %g, avg-rtt %" G_GUINT64_FORMAT, priv->num_rtx_success,
priv->num_rtx_failed, priv->num_rtx_requests, priv->num_duplicates,
priv->avg_rtx_num, priv->avg_rtx_rtt);
}
/* if we had a timer, remove it, we don't know when to expect the next
* packet. */
remove_timer (jitterbuffer, timer);
/* we signal the _loop function because this new packet could be the one
* it was waiting for */
JBUF_SIGNAL_EVENT (priv);
}
}
static void
calculate_packet_spacing (GstRtpJitterBuffer * jitterbuffer, guint32 rtptime,
GstClockTime dts)
{
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 dts diff */
if (priv->ips_dts != -1 && dts != -1 && dts > priv->ips_dts) {
priv->packet_spacing = dts - priv->ips_dts;
GST_DEBUG_OBJECT (jitterbuffer,
"new packet spacing %" GST_TIME_FORMAT,
GST_TIME_ARGS (priv->packet_spacing));
}
priv->ips_rtptime = rtptime;
priv->ips_dts = dts;
}
}
static void
calculate_expected (GstRtpJitterBuffer * jitterbuffer, guint32 expected,
guint16 seqnum, GstClockTime dts, gint gap)
{
GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
GstClockTime total_duration, duration, expected_dts;
TimerType type;
GST_DEBUG_OBJECT (jitterbuffer,
"dts %" GST_TIME_FORMAT ", last %" GST_TIME_FORMAT,
GST_TIME_ARGS (dts), GST_TIME_ARGS (priv->last_in_dts));
/* the total duration spanned by the missing packets */
if (dts >= priv->last_in_dts)
total_duration = dts - priv->last_in_dts;
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;
gap_time = total_duration - priv->latency_ns;
if (duration > 0) {
lost_packets = gap_time / duration;
gap_time = lost_packets * duration;
} else {
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_DEBUG_OBJECT (jitterbuffer, "too many lost packets %" GST_TIME_FORMAT
" > %" GST_TIME_FORMAT ", consider %u lost",
GST_TIME_ARGS (total_duration), GST_TIME_ARGS (priv->latency_ns),
lost_packets);
/* this timer will fire immediately and the lost event will be pushed from
* the timer thread */
add_timer (jitterbuffer, TIMER_TYPE_LOST, expected, lost_packets,
priv->last_in_dts + duration, 0, gap_time);
expected += lost_packets;
priv->last_in_dts += gap_time;
}
expected_dts = priv->last_in_dts + duration;
if (priv->do_retransmission) {
TimerData *timer;
type = TIMER_TYPE_EXPECTED;
/* if we had a timer for the first missing packet, update it. */
if ((timer = find_timer (jitterbuffer, type, expected))) {
GstClockTime timeout = timer->timeout;
timer->duration = duration;
if (timeout > expected_dts) {
GstClockTime delay = timeout - expected_dts - timer->rtx_retry;
reschedule_timer (jitterbuffer, timer, timer->seqnum, expected_dts,
delay, TRUE);
}
expected++;
expected_dts += duration;
}
} else {
type = TIMER_TYPE_LOST;
}
while (expected < seqnum) {
add_timer (jitterbuffer, type, expected, 0, expected_dts, 0, duration);
expected_dts += duration;
expected++;
}
}
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 tail;
gint percent = -1;
guint8 pt;
GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
gboolean do_next_seqnum = FALSE;
RTPJitterBufferItem *item;
jitterbuffer = GST_RTP_JITTER_BUFFER (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);
/* 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;
/* 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", seqnum,
GST_TIME_ARGS (dts), GST_BUFFER_IS_DISCONT (buffer));
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);
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;
}
/* don't accept more data on EOS */
if (G_UNLIKELY (priv->eos))
goto have_eos;
expected = priv->next_in_seqnum;
/* now check against our expected seqnum */
if (G_LIKELY (expected != -1)) {
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_LIKELY (gap == 0)) {
/* packet is expected */
calculate_packet_spacing (jitterbuffer, rtptime, dts);
do_next_seqnum = TRUE;
} else {
gboolean reset = FALSE;
if (gap < 0) {
/* we received an old packet */
if (G_UNLIKELY (gap < -RTP_MAX_MISORDER)) {
/* too old packet, reset */
GST_DEBUG_OBJECT (jitterbuffer, "reset: buffer too old %d < %d", gap,
-RTP_MAX_MISORDER);
reset = TRUE;
} else {
GST_DEBUG_OBJECT (jitterbuffer, "old packet received");
}
} else {
/* new packet, we are missing some packets */
if (G_UNLIKELY (gap > RTP_MAX_DROPOUT)) {
/* packet too far in future, reset */
GST_DEBUG_OBJECT (jitterbuffer, "reset: buffer too new %d > %d", gap,
RTP_MAX_DROPOUT);
reset = TRUE;
} else {
GST_DEBUG_OBJECT (jitterbuffer, "%d missing packets", gap);
/* fill in the gap with EXPECTED timers */
calculate_expected (jitterbuffer, expected, seqnum, dts, gap);
do_next_seqnum = TRUE;
}
}
if (G_UNLIKELY (reset)) {
GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
rtp_jitter_buffer_flush (priv->jbuf, (GFunc) free_item, NULL);
rtp_jitter_buffer_reset_skew (priv->jbuf);
remove_all_timers (jitterbuffer);
priv->last_popped_seqnum = -1;
priv->next_seqnum = seqnum;
do_next_seqnum = TRUE;
}
/* reset spacing estimation when gap */
priv->ips_rtptime = -1;
priv->ips_dts = GST_CLOCK_TIME_NONE;
}
} else {
GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum);
/* 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 */
set_timer (jitterbuffer, TIMER_TYPE_DEADLINE, seqnum, dts);
do_next_seqnum = TRUE;
/* take rtptime and dts to calculate packet spacing */
priv->ips_rtptime = rtptime;
priv->ips_dts = dts;
}
if (do_next_seqnum) {
priv->last_in_seqnum = seqnum;
priv->last_in_dts = dts;
priv->next_in_seqnum = (seqnum + 1) & 0xffff;
}
/* let's check if this buffer is too late, we can only accept packets with
* bigger seqnum than the one we last pushed. */
if (G_LIKELY (priv->last_popped_seqnum != -1)) {
gint gap;
gap = gst_rtp_buffer_compare_seqnum (priv->last_popped_seqnum, seqnum);
/* priv->last_popped_seqnum >= seqnum, we're too late. */
if (G_UNLIKELY (gap <= 0))
goto too_late;
}
/* let's drop oldest packet if the queue is already full and drop-on-latency
* is set. We can only do this when there actually is a latency. When no
* latency is set, we just pump it in the queue and let the other end push it
* out as fast as possible. */
if (priv->latency_ms && priv->drop_on_latency) {
latency_ts =
gst_util_uint64_scale_int (priv->latency_ms, priv->clock_rate, 1000);
if (G_UNLIKELY (rtp_jitter_buffer_get_ts_diff (priv->jbuf) >= latency_ts)) {
RTPJitterBufferItem *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 + 1) & 0xffff;
free_item (old_item);
}
}
item = alloc_item (buffer, ITEM_TYPE_BUFFER, dts, pts, seqnum, 1, rtptime);
/* now insert the packet into the queue in sorted order. This function returns
* FALSE if a packet with the same seqnum was already in the queue, meaning we
* have a duplicate. */
if (G_UNLIKELY (!rtp_jitter_buffer_insert (priv->jbuf, item,
&tail, &percent)))
goto duplicate;
/* update timers */
update_timers (jitterbuffer, seqnum, dts, do_next_seqnum);
/* we had an unhandled SR, handle it now */
if (priv->last_sr)
do_handle_sync (jitterbuffer);
/* signal addition of new buffer when the _loop is waiting. */
if (priv->active)
JBUF_SIGNAL_EVENT (priv);
/* let's unschedule and unblock any waiting buffers. We only want to do this
* when the tail buffer changed */
if (G_UNLIKELY (priv->clock_id && tail)) {
GST_DEBUG_OBJECT (jitterbuffer, "Unscheduling waiting new buffer");
unschedule_current_timer (jitterbuffer);
}
GST_DEBUG_OBJECT (jitterbuffer, "Pushed packet #%d, now %d packets, tail: %d",
seqnum, rtp_jitter_buffer_num_packets (priv->jbuf), tail);
check_buffering_percent (jitterbuffer, &percent);
finished:
JBUF_UNLOCK (priv);
if (percent != -1)
post_buffering_percent (jitterbuffer, percent);
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_WARNING_OBJECT (jitterbuffer, "Packet #%d too late as #%d was already"
" popped, dropping", seqnum, priv->last_popped_seqnum);
priv->num_late++;
gst_buffer_unref (buffer);
goto finished;
}
duplicate:
{
GST_WARNING_OBJECT (jitterbuffer, "Duplicate packet #%d detected, dropping",
seqnum);
priv->num_duplicates++;
free_item (item);
goto finished;
}
}
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);
if (rtp_time < priv->ext_timestamp) {
ext_time = priv->ext_timestamp;
} else {
ext_time = gst_rtp_buffer_ext_timestamp (&priv->ext_timestamp, rtp_time);
}
if (ext_time > priv->clock_base)
elapsed = ext_time - priv->clock_base;
else
elapsed = 0;
elapsed = gst_util_uint64_scale_int (elapsed, GST_SECOND, priv->clock_rate);
return elapsed;
}
static void
update_estimated_eos (GstRtpJitterBuffer * jitterbuffer,
RTPJitterBufferItem * item)
{
GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
if (priv->npt_stop != -1 && priv->ext_timestamp != -1
&& priv->clock_base != -1 && priv->clock_rate > 0) {
guint64 elapsed, estimated;
elapsed = compute_elapsed (jitterbuffer, item);
if (elapsed > priv->last_elapsed || !priv->last_elapsed) {
guint64 left;
GstClockTime out_time;
priv->last_elapsed = elapsed;
left = priv->npt_stop - priv->npt_start;
GST_LOG_OBJECT (jitterbuffer, "left %" GST_TIME_FORMAT,
GST_TIME_ARGS (left));
out_time = item->dts;
if (elapsed > 0)
estimated = gst_util_uint64_scale (out_time, left, elapsed);
else {
/* if there is almost nothing left,
* we may never advance enough to end up in the above case */
if (left < GST_SECOND)
estimated = GST_SECOND;
else
estimated = -1;
}
GST_LOG_OBJECT (jitterbuffer, "elapsed %" GST_TIME_FORMAT ", estimated %"
GST_TIME_FORMAT, GST_TIME_ARGS (elapsed), GST_TIME_ARGS (estimated));
if (estimated != -1 && priv->estimated_eos != estimated) {
set_timer (jitterbuffer, TIMER_TYPE_EOS, -1, estimated);
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;
RTPJitterBufferItem *item;
GstBuffer *outbuf;
GstEvent *outevent;
GstClockTime dts, pts;
gint percent = -1;
gboolean is_buffer, do_push = TRUE;
/* when we get here we are ready to pop and push the buffer */
item = rtp_jitter_buffer_pop (priv->jbuf, &percent);
is_buffer = GST_IS_BUFFER (item->data);
if (is_buffer) {
check_buffering_percent (jitterbuffer, &percent);
/* 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_to_position (&priv->segment, GST_FORMAT_TIME, item->dts);
pts = gst_segment_to_position (&priv->segment, GST_FORMAT_TIME, item->pts);
/* 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_out_time = GST_BUFFER_PTS (outbuf);
} else {
outevent = item->data;
if (item->type == ITEM_TYPE_LOST) {
priv->discont = TRUE;
if (!priv->do_lost)
do_push = FALSE;
}
}
/* 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;
}
JBUF_UNLOCK (priv);
item->data = NULL;
free_item (item);
if (is_buffer) {
/* push buffer */
if (percent != -1)
post_buffering_percent (jitterbuffer, percent);
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)));
result = gst_pad_push (priv->srcpad, outbuf);
} else {
GST_DEBUG_OBJECT (jitterbuffer, "Pushing event %d", seqnum);
if (do_push)
gst_pad_push_event (priv->srcpad, outevent);
else
gst_event_unref (outevent);
result = GST_FLOW_OK;
}
JBUF_LOCK_CHECK (priv, out_flushing);
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 = GST_FLOW_OK;
RTPJitterBufferItem *item;
guint seqnum;
guint32 next_seqnum;
gint gap;
/* 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;
again:
/* 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)
goto do_push;
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 {
/* else calculate GAP */
gap = gst_rtp_buffer_compare_seqnum (next_seqnum, seqnum);
if (G_LIKELY (gap == 0)) {
do_push:
/* no missing packet, pop and push */
result = pop_and_push_next (jitterbuffer, seqnum);
} else if (G_UNLIKELY (gap < 0)) {
RTPJitterBufferItem *item;
/* 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);
free_item (item);
goto again;
} 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);
result = GST_FLOW_WAIT;
}
}
return result;
wait:
{
GST_DEBUG_OBJECT (jitterbuffer, "no buffer, going to wait");
if (priv->eos)
result = GST_FLOW_EOS;
else
result = GST_FLOW_WAIT;
return result;
}
}
/* the timeout for when we expected a packet expired */
static gboolean
do_expected_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
GstClockTime now)
{
GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
GstEvent *event;
guint delay;
GST_DEBUG_OBJECT (jitterbuffer, "expected %d didn't arrive", timer->seqnum);
delay = timer->rtx_delay + timer->rtx_retry;
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, GST_TIME_AS_MSECONDS (delay),
"retry", G_TYPE_UINT, timer->num_rtx_retry,
"frequency", G_TYPE_UINT, priv->rtx_retry_timeout,
"period", G_TYPE_UINT, priv->rtx_retry_period,
"deadline", G_TYPE_UINT, priv->latency_ms,
"packet-spacing", G_TYPE_UINT64, priv->packet_spacing, NULL));
priv->num_rtx_requests++;
timer->num_rtx_retry++;
timer->rtx_last = now;
/* calculate the timeout for the next retransmission attempt */
timer->rtx_retry += (priv->rtx_retry_timeout * GST_MSECOND);
GST_DEBUG_OBJECT (jitterbuffer, "base %" GST_TIME_FORMAT ", delay %"
GST_TIME_FORMAT ", retry %" GST_TIME_FORMAT,
GST_TIME_ARGS (timer->rtx_base), GST_TIME_ARGS (timer->rtx_delay),
GST_TIME_ARGS (timer->rtx_retry));
if (timer->rtx_retry + timer->rtx_delay >
(priv->rtx_retry_period * GST_MSECOND)) {
GST_DEBUG_OBJECT (jitterbuffer, "reschedule as LOST timer");
/* 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 = TIMER_TYPE_LOST;
timer->rtx_delay = 0;
timer->rtx_retry = 0;
}
reschedule_timer (jitterbuffer, timer, timer->seqnum,
timer->rtx_base + timer->rtx_retry, timer->rtx_delay, 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, TimerData * timer,
GstClockTime now)
{
GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
GstClockTime duration, timestamp;
guint seqnum, lost_packets, num_rtx_retry;
gboolean late;
GstEvent *event;
RTPJitterBufferItem *item;
seqnum = timer->seqnum;
timestamp = apply_offset (jitterbuffer, timer->timeout);
duration = timer->duration;
if (duration == GST_CLOCK_TIME_NONE && priv->packet_spacing > 0)
duration = priv->packet_spacing;
lost_packets = MAX (timer->num, 1);
late = timer->num > 0;
num_rtx_retry = timer->num_rtx_retry;
/* 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_late += lost_packets;
priv->num_rtx_failed += num_rtx_retry;
/* create paket lost event */
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,
"late", G_TYPE_BOOLEAN, late,
"retry", G_TYPE_UINT, num_rtx_retry, NULL));
item = alloc_item (event, ITEM_TYPE_LOST, -1, -1, seqnum, lost_packets, -1);
rtp_jitter_buffer_insert (priv->jbuf, item, NULL, NULL);
/* remove timer now */
remove_timer (jitterbuffer, timer);
JBUF_SIGNAL_EVENT (priv);
return TRUE;
}
static gboolean
do_eos_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
GstClockTime now)
{
GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
GST_INFO_OBJECT (jitterbuffer, "got the NPT timeout");
remove_timer (jitterbuffer, timer);
JBUF_SIGNAL_EVENT (priv);
return TRUE;
}
static gboolean
do_deadline_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
GstClockTime now)
{
GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
GST_INFO_OBJECT (jitterbuffer, "got deadline timeout");
priv->next_seqnum = timer->seqnum;
remove_timer (jitterbuffer, timer);
JBUF_SIGNAL_EVENT (priv);
return TRUE;
}
static gboolean
do_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
GstClockTime now)
{
gboolean removed = FALSE;
switch (timer->type) {
case TIMER_TYPE_EXPECTED:
removed = do_expected_timeout (jitterbuffer, timer, now);
break;
case TIMER_TYPE_LOST:
removed = do_lost_timeout (jitterbuffer, timer, now);
break;
case TIMER_TYPE_DEADLINE:
removed = do_deadline_timeout (jitterbuffer, timer, now);
break;
case TIMER_TYPE_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) {
TimerData *timer = NULL;
GstClockTime timer_timeout = -1;
gint i, len;
GST_DEBUG_OBJECT (jitterbuffer, "now %" GST_TIME_FORMAT,
GST_TIME_ARGS (now));
len = priv->timers->len;
for (i = 0; i < len; i++) {
TimerData *test = &g_array_index (priv->timers, TimerData, i);
GstClockTime test_timeout = get_timeout (jitterbuffer, test);
gboolean save_best = FALSE;
GST_DEBUG_OBJECT (jitterbuffer, "%d, %d, %d, %" GST_TIME_FORMAT,
i, test->type, test->seqnum, GST_TIME_ARGS (test_timeout));
/* find the smallest timeout */
if (timer == NULL) {
save_best = TRUE;
} else if (timer_timeout == -1) {
/* we already have an immediate timeout, the new timer must be an
* immediate timer with smaller seqnum to become the best */
if (test_timeout == -1 && test->seqnum < timer->seqnum)
save_best = TRUE;
} else if (test_timeout == -1) {
/* first immediate timer */
save_best = TRUE;
} else if (test_timeout < timer_timeout) {
/* earlier timer */
save_best = TRUE;
} else if (test_timeout == timer_timeout && test->seqnum < timer->seqnum) {
/* same timer, smaller seqnum */
save_best = TRUE;
}
if (save_best) {
GST_DEBUG_OBJECT (jitterbuffer, "new best %d", i);
timer = test;
timer_timeout = test_timeout;
}
}
if (timer && !priv->blocked) {
GstClock *clock;
GstClockTime sync_time;
GstClockID id;
GstClockReturn ret;
GstClockTimeDiff clock_jitter;
if (timer_timeout == -1 || timer_timeout <= now) {
do_timeout (jitterbuffer, timer, now);
/* check here, do_timeout could have released the lock */
if (!priv->timer_running)
break;
continue;
}
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, "sync to timestamp %" GST_TIME_FORMAT
" with sync time %" GST_TIME_FORMAT,
GST_TIME_ARGS (timer_timeout), 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)
break;
if (ret != GST_CLOCK_UNSCHEDULED) {
now = timer_timeout + MAX (clock_jitter, 0);
GST_DEBUG_OBJECT (jitterbuffer, "sync done, %d, #%d, %" G_GINT64_FORMAT,
ret, priv->timer_seqnum, clock_jitter);
} else {
GST_DEBUG_OBJECT (jitterbuffer, "sync unscheduled");
}
/* and free the entry */
gst_clock_id_unref (id);
priv->clock_id = NULL;
} else {
/* no timers, wait for activity */
JBUF_WAIT_TIMER (priv);
}
}
JBUF_UNLOCK (priv);
GST_DEBUG_OBJECT (jitterbuffer, "we are stopping");
return;
}
/*
* This funcion 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;
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_WAIT_EVENT (priv, flushing);
result = GST_FLOW_OK;
}
}
while (result == GST_FLOW_OK);
/* store result for upstream */
priv->srcresult = result;
JBUF_UNLOCK (priv);
/* if we get here we need to pause */
goto pause;
/* ERRORS */
flushing:
{
result = priv->srcresult;
JBUF_UNLOCK (priv);
goto pause;
}
pause:
{
const gchar *reason = gst_flow_get_name (result);
GstEvent *event;
GST_DEBUG_OBJECT (jitterbuffer, "pausing task, reason %s", reason);
gst_pad_pause_task (priv->srcpad);
if (result == GST_FLOW_EOS) {
event = gst_event_new_eos ();
gst_pad_push_event (priv->srcpad, event);
}
return;
}
}
/* collect the info from the lastest 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 drop = 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) {
GST_DEBUG_OBJECT (jitterbuffer, "dropping, no RTP values");
drop = TRUE;
} else {
/* we can't accept anything that happened before we did the last resync */
if (base_rtptime > ext_rtptime) {
GST_DEBUG_OBJECT (jitterbuffer, "dropping, older than base time");
drop = TRUE;
} else {
/* the SR RTP timestamp must be something close to what we last observed
* in the jitterbuffer */
if (ext_rtptime > last_rtptime) {
/* check how far ahead it is to our RTP timestamps */
diff = ext_rtptime - last_rtptime;
/* if bigger than 1 second, we drop it */
if (diff > clock_rate) {
GST_DEBUG_OBJECT (jitterbuffer, "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 (!drop) {
GstStructure *s;
s = gst_structure_new ("application/x-rtp-sync",
"base-rtptime", G_TYPE_UINT64, base_rtptime,
"base-time", G_TYPE_UINT64, base_time,
"clock-rate", G_TYPE_UINT, clock_rate,
"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");
}
}
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 (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;
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)) {
GST_WARNING_OBJECT (pad, "unhandled serialized query");
res = FALSE;
} else {
res = gst_pad_query_default (pad, parent, query);
}
break;
}
return res;
}
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);
priv->ts_offset = g_value_get_int64 (value);
priv->ts_discont = TRUE;
JBUF_UNLOCK (priv);
break;
case PROP_DO_LOST:
JBUF_LOCK (priv);
priv->do_lost = g_value_get_boolean (value);
JBUF_UNLOCK (priv);
break;
case PROP_MODE:
JBUF_LOCK (priv);
rtp_jitter_buffer_set_mode (priv->jbuf, g_value_get_enum (value));
JBUF_UNLOCK (priv);
break;
case PROP_DO_RETRANSMISSION:
JBUF_LOCK (priv);
priv->do_retransmission = 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_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_RETRY_PERIOD:
JBUF_LOCK (priv);
priv->rtx_retry_period = g_value_get_int (value);
JBUF_UNLOCK (priv);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_rtp_jitter_buffer_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec)
{
GstRtpJitterBuffer *jitterbuffer;
GstRtpJitterBufferPrivate *priv;
jitterbuffer = GST_RTP_JITTER_BUFFER (object);
priv = jitterbuffer->priv;
switch (prop_id) {
case PROP_LATENCY:
JBUF_LOCK (priv);
g_value_set_uint (value, priv->latency_ms);
JBUF_UNLOCK (priv);
break;
case PROP_DROP_ON_LATENCY:
JBUF_LOCK (priv);
g_value_set_boolean (value, priv->drop_on_latency);
JBUF_UNLOCK (priv);
break;
case PROP_TS_OFFSET:
JBUF_LOCK (priv);
g_value_set_int64 (value, priv->ts_offset);
JBUF_UNLOCK (priv);
break;
case PROP_DO_LOST:
JBUF_LOCK (priv);
g_value_set_boolean (value, priv->do_lost);
JBUF_UNLOCK (priv);
break;
case PROP_MODE:
JBUF_LOCK (priv);
g_value_set_enum (value, rtp_jitter_buffer_get_mode (priv->jbuf));
JBUF_UNLOCK (priv);
break;
case PROP_PERCENT:
{
gint percent;
JBUF_LOCK (priv);
if (priv->srcresult != GST_FLOW_OK)
percent = 100;
else
percent = rtp_jitter_buffer_get_percent (priv->jbuf);
g_value_set_int (value, percent);
JBUF_UNLOCK (priv);
break;
}
case PROP_DO_RETRANSMISSION:
JBUF_LOCK (priv);
g_value_set_boolean (value, priv->do_retransmission);
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_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_RETRY_PERIOD:
JBUF_LOCK (priv);
g_value_set_int (value, priv->rtx_retry_period);
JBUF_UNLOCK (priv);
break;
case PROP_STATS:
g_value_take_boxed (value,
gst_rtp_jitter_buffer_create_stats (jitterbuffer));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstStructure *
gst_rtp_jitter_buffer_create_stats (GstRtpJitterBuffer * jbuf)
{
GstStructure *s;
JBUF_LOCK (jbuf->priv);
s = gst_structure_new ("application/x-rtp-jitterbuffer-stats",
"rtx-count", G_TYPE_UINT64, jbuf->priv->num_rtx_requests,
"rtx-success-count", G_TYPE_UINT64, jbuf->priv->num_rtx_success,
"rtx-per-packet", G_TYPE_DOUBLE, jbuf->priv->avg_rtx_num,
"rtx-rtt", G_TYPE_UINT64, jbuf->priv->avg_rtx_rtt, NULL);
JBUF_UNLOCK (jbuf->priv);
return s;
}
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