mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-28 04:31:06 +00:00
87c8c163a8
We (currently?) can't really handle gaps between RTP packets if they're not properly timestamped. The current code would go into calculations with GST_CLOCK_TIME_NONE and then cause assertions everywhere. It's probably better to error out cleanly instead.
3678 lines
113 KiB
C
3678 lines
113 KiB
C
/*
|
|
* 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>
|
|
*/
|
|
|
|
#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 -1
|
|
#define DEFAULT_RTX_MIN_DELAY 0
|
|
#define DEFAULT_RTX_DELAY_REORDER 3
|
|
#define DEFAULT_RTX_RETRY_TIMEOUT -1
|
|
#define DEFAULT_RTX_MIN_RETRY_TIMEOUT -1
|
|
#define DEFAULT_RTX_RETRY_PERIOD -1
|
|
|
|
#define DEFAULT_AUTO_RTX_DELAY (20 * GST_MSECOND)
|
|
#define DEFAULT_AUTO_RTX_TIMEOUT (40 * GST_MSECOND)
|
|
|
|
enum
|
|
{
|
|
PROP_0,
|
|
PROP_LATENCY,
|
|
PROP_DROP_ON_LATENCY,
|
|
PROP_TS_OFFSET,
|
|
PROP_DO_LOST,
|
|
PROP_MODE,
|
|
PROP_PERCENT,
|
|
PROP_DO_RETRANSMISSION,
|
|
PROP_RTX_DELAY,
|
|
PROP_RTX_MIN_DELAY,
|
|
PROP_RTX_DELAY_REORDER,
|
|
PROP_RTX_RETRY_TIMEOUT,
|
|
PROP_RTX_MIN_RETRY_TIMEOUT,
|
|
PROP_RTX_RETRY_PERIOD,
|
|
PROP_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
|
|
|
|
#define JBUF_WAIT_QUERY(priv,label) G_STMT_START { \
|
|
GST_DEBUG ("waiting query"); \
|
|
(priv)->waiting_query = TRUE; \
|
|
g_cond_wait (&(priv)->jbuf_query, &(priv)->jbuf_lock); \
|
|
(priv)->waiting_query = FALSE; \
|
|
GST_DEBUG ("waiting query done"); \
|
|
if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
|
|
goto label; \
|
|
} G_STMT_END
|
|
#define JBUF_SIGNAL_QUERY(priv,res) G_STMT_START { \
|
|
(priv)->last_query = res; \
|
|
if (G_UNLIKELY ((priv)->waiting_query)) { \
|
|
GST_DEBUG ("signal query"); \
|
|
g_cond_signal (&(priv)->jbuf_query); \
|
|
} \
|
|
} G_STMT_END
|
|
|
|
|
|
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 waiting_query;
|
|
GCond jbuf_query;
|
|
gboolean last_query;
|
|
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;
|
|
guint rtx_min_delay;
|
|
gint rtx_delay_reorder;
|
|
gint rtx_retry_timeout;
|
|
gint rtx_min_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;
|
|
guint last_percent;
|
|
|
|
/* 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;
|
|
|
|
/* for the jitter */
|
|
GstClockTime last_dts;
|
|
guint64 last_rtptime;
|
|
GstClockTime avg_jitter;
|
|
};
|
|
|
|
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-min-delay:
|
|
*
|
|
* When a packet did not arrive at the expected time, wait at least this extra amount
|
|
* of time before sending a retransmission event.
|
|
*
|
|
* Since: 1.6
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_RTX_MIN_DELAY,
|
|
g_param_spec_uint ("rtx-min-delay", "Minimum RTX Delay",
|
|
"Minimum time in ms to wait before sending retransmission "
|
|
"event", 0, G_MAXUINT, DEFAULT_RTX_MIN_DELAY,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
/**
|
|
* GstRtpJitterBuffer:rtx-delay-reorder:
|
|
*
|
|
* Assume that a retransmission event should be sent when we see
|
|
* this much packet reordering.
|
|
*
|
|
* When -1 is used, the value will be estimated based on observed packet
|
|
* reordering.
|
|
*
|
|
* 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-min-retry-timeout:
|
|
*
|
|
* The minimum amount of time between retry timeouts. When
|
|
* GstRtpJitterBuffer::rtx-retry-timeout is -1, this value ensures a
|
|
* minimum interval between retry timeouts.
|
|
*
|
|
* When -1 is used, the value will be estimated based on the
|
|
* packet spacing.
|
|
*
|
|
* Since: 1.6
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_RTX_MIN_RETRY_TIMEOUT,
|
|
g_param_spec_int ("rtx-min-retry-timeout", "RTX Min Retry Timeout",
|
|
"Minimum timeout between sending a transmission event in "
|
|
"ms (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_MIN_RETRY_TIMEOUT,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
/**
|
|
* GstRtpJitterBuffer:rtx-retry-period:
|
|
*
|
|
* The amount of time to try to get a retransmission.
|
|
*
|
|
* When -1 is used, the value will be estimated based on the jitterbuffer
|
|
* latency and the observed round trip time.
|
|
*
|
|
* Since: 1.2
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_RTX_RETRY_PERIOD,
|
|
g_param_spec_int ("rtx-retry-period", "RTX Retry Period",
|
|
"Try to get a retransmission for this many ms "
|
|
"(-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_RETRY_PERIOD,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
/**
|
|
* GstRtpJitterBuffer: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_min_delay = DEFAULT_RTX_MIN_DELAY;
|
|
priv->rtx_delay_reorder = DEFAULT_RTX_DELAY_REORDER;
|
|
priv->rtx_retry_timeout = DEFAULT_RTX_RETRY_TIMEOUT;
|
|
priv->rtx_min_retry_timeout = DEFAULT_RTX_MIN_RETRY_TIMEOUT;
|
|
priv->rtx_retry_period = DEFAULT_RTX_RETRY_PERIOD;
|
|
|
|
priv->last_dts = -1;
|
|
priv->last_rtptime = -1;
|
|
priv->avg_jitter = 0;
|
|
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);
|
|
g_cond_init (&priv->jbuf_query);
|
|
|
|
/* 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 IS_DROPABLE(it) (((it)->type == ITEM_TYPE_BUFFER) || ((it)->type == ITEM_TYPE_LOST))
|
|
|
|
#define ITEM_TYPE_BUFFER 0
|
|
#define ITEM_TYPE_LOST 1
|
|
#define ITEM_TYPE_EVENT 2
|
|
#define ITEM_TYPE_QUERY 3
|
|
|
|
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 && item->type != ITEM_TYPE_QUERY)
|
|
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);
|
|
g_cond_clear (&priv->jbuf_query);
|
|
|
|
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 = NULL;
|
|
GValue val = { 0, };
|
|
|
|
jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
|
|
|
|
if (pad == jitterbuffer->priv->sinkpad) {
|
|
otherpad = jitterbuffer->priv->srcpad;
|
|
} else if (pad == jitterbuffer->priv->srcpad) {
|
|
otherpad = jitterbuffer->priv->sinkpad;
|
|
} else if (pad == jitterbuffer->priv->rtcpsinkpad) {
|
|
it = gst_iterator_new_single (GST_TYPE_PAD, NULL);
|
|
}
|
|
|
|
if (it == NULL) {
|
|
g_value_init (&val, GST_TYPE_PAD);
|
|
g_value_set_object (&val, otherpad);
|
|
it = gst_iterator_new_single (GST_TYPE_PAD, &val);
|
|
g_value_unset (&val);
|
|
}
|
|
|
|
return it;
|
|
}
|
|
|
|
static GstPad *
|
|
create_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv;
|
|
|
|
priv = jitterbuffer->priv;
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "creating RTCP sink pad");
|
|
|
|
priv->rtcpsinkpad =
|
|
gst_pad_new_from_static_template
|
|
(&gst_rtp_jitter_buffer_sink_rtcp_template, "sink_rtcp");
|
|
gst_pad_set_chain_function (priv->rtcpsinkpad,
|
|
gst_rtp_jitter_buffer_chain_rtcp);
|
|
gst_pad_set_event_function (priv->rtcpsinkpad,
|
|
(GstPadEventFunction) gst_rtp_jitter_buffer_sink_rtcp_event);
|
|
gst_pad_set_iterate_internal_links_function (priv->rtcpsinkpad,
|
|
gst_rtp_jitter_buffer_iterate_internal_links);
|
|
gst_pad_set_active (priv->rtcpsinkpad, TRUE);
|
|
gst_element_add_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
|
|
|
|
return priv->rtcpsinkpad;
|
|
}
|
|
|
|
static void
|
|
remove_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv;
|
|
|
|
priv = jitterbuffer->priv;
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "removing RTCP sink pad");
|
|
|
|
gst_pad_set_active (priv->rtcpsinkpad, FALSE);
|
|
|
|
gst_element_remove_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
|
|
priv->rtcpsinkpad = NULL;
|
|
}
|
|
|
|
static GstPad *
|
|
gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
|
|
GstPadTemplate * templ, const gchar * name, const GstCaps * filter)
|
|
{
|
|
GstRtpJitterBuffer *jitterbuffer;
|
|
GstElementClass *klass;
|
|
GstPad *result;
|
|
GstRtpJitterBufferPrivate *priv;
|
|
|
|
g_return_val_if_fail (templ != NULL, NULL);
|
|
g_return_val_if_fail (GST_IS_RTP_JITTER_BUFFER (element), NULL);
|
|
|
|
jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (element);
|
|
priv = jitterbuffer->priv;
|
|
klass = GST_ELEMENT_GET_CLASS (element);
|
|
|
|
GST_DEBUG_OBJECT (element, "requesting pad %s", GST_STR_NULL (name));
|
|
|
|
/* figure out the template */
|
|
if (templ == gst_element_class_get_pad_template (klass, "sink_rtcp")) {
|
|
if (priv->rtcpsinkpad != NULL)
|
|
goto exists;
|
|
|
|
result = create_rtcp_sink (jitterbuffer);
|
|
} else
|
|
goto wrong_template;
|
|
|
|
return result;
|
|
|
|
/* ERRORS */
|
|
wrong_template:
|
|
{
|
|
g_warning ("rtpjitterbuffer: this is not our template");
|
|
return NULL;
|
|
}
|
|
exists:
|
|
{
|
|
g_warning ("rtpjitterbuffer: pad already requested");
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtp_jitter_buffer_release_pad (GstElement * element, GstPad * pad)
|
|
{
|
|
GstRtpJitterBuffer *jitterbuffer;
|
|
GstRtpJitterBufferPrivate *priv;
|
|
|
|
g_return_if_fail (GST_IS_RTP_JITTER_BUFFER (element));
|
|
g_return_if_fail (GST_IS_PAD (pad));
|
|
|
|
jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (element);
|
|
priv = jitterbuffer->priv;
|
|
|
|
GST_DEBUG_OBJECT (element, "releasing pad %s:%s", GST_DEBUG_PAD_NAME (pad));
|
|
|
|
if (priv->rtcpsinkpad == pad) {
|
|
remove_rtcp_sink (jitterbuffer);
|
|
} else
|
|
goto wrong_pad;
|
|
|
|
return;
|
|
|
|
/* ERRORS */
|
|
wrong_pad:
|
|
{
|
|
g_warning ("gstjitterbuffer: asked to release an unknown pad");
|
|
return;
|
|
}
|
|
}
|
|
|
|
static GstClock *
|
|
gst_rtp_jitter_buffer_provide_clock (GstElement * element)
|
|
{
|
|
return gst_system_clock_obtain ();
|
|
}
|
|
|
|
static 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;
|
|
JBUF_SIGNAL_EVENT (priv);
|
|
}
|
|
}
|
|
|
|
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_SIGNAL_QUERY (priv, FALSE);
|
|
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->last_pt = -1;
|
|
priv->eos = FALSE;
|
|
priv->estimated_eos = -1;
|
|
priv->last_elapsed = 0;
|
|
priv->ext_timestamp = -1;
|
|
priv->avg_jitter = 0;
|
|
priv->last_dts = -1;
|
|
priv->last_rtptime = -1;
|
|
GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
|
|
rtp_jitter_buffer_flush (priv->jbuf, (GFunc) free_item, NULL);
|
|
rtp_jitter_buffer_disable_buffering (priv->jbuf, FALSE);
|
|
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_SIGNAL_QUERY (priv, FALSE);
|
|
JBUF_UNLOCK (priv);
|
|
g_thread_join (priv->timer_thread);
|
|
priv->timer_thread = NULL;
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_NULL:
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_jitter_buffer_src_event (GstPad * pad, GstObject * parent,
|
|
GstEvent * event)
|
|
{
|
|
gboolean ret = TRUE;
|
|
GstRtpJitterBuffer *jitterbuffer;
|
|
GstRtpJitterBufferPrivate *priv;
|
|
|
|
jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
|
|
priv = jitterbuffer->priv;
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_LATENCY:
|
|
{
|
|
GstClockTime latency;
|
|
|
|
gst_event_parse_latency (event, &latency);
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer,
|
|
"configuring latency of %" GST_TIME_FORMAT, GST_TIME_ARGS (latency));
|
|
|
|
JBUF_LOCK (priv);
|
|
/* adjust the overall buffer delay to the total pipeline latency in
|
|
* buffering mode because if downstream consumes too fast (because of
|
|
* large latency or queues, we would start rebuffering again. */
|
|
if (rtp_jitter_buffer_get_mode (priv->jbuf) ==
|
|
RTP_JITTER_BUFFER_MODE_BUFFER) {
|
|
rtp_jitter_buffer_set_delay (priv->jbuf, latency);
|
|
}
|
|
JBUF_UNLOCK (priv);
|
|
|
|
ret = gst_pad_push_event (priv->sinkpad, event);
|
|
break;
|
|
}
|
|
default:
|
|
ret = gst_pad_push_event (priv->sinkpad, event);
|
|
break;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
/* handles and stores the event in the jitterbuffer, must be called with
|
|
* LOCK */
|
|
static gboolean
|
|
queue_event (GstRtpJitterBuffer * jitterbuffer, GstEvent * event)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
|
|
RTPJitterBufferItem *item;
|
|
gboolean head;
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_CAPS:
|
|
{
|
|
GstCaps *caps;
|
|
|
|
gst_event_parse_caps (event, &caps);
|
|
gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
|
|
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;
|
|
rtp_jitter_buffer_disable_buffering (priv->jbuf, 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, &head, NULL);
|
|
if (head)
|
|
JBUF_SIGNAL_EVENT (priv);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
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 GstMessage *
|
|
check_buffering_percent (GstRtpJitterBuffer * jitterbuffer, gint percent)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
|
|
GstMessage *message = NULL;
|
|
|
|
if (percent == -1)
|
|
return NULL;
|
|
|
|
/* Post a buffering message */
|
|
if (priv->last_percent != percent) {
|
|
priv->last_percent = percent;
|
|
message =
|
|
gst_message_new_buffering (GST_OBJECT_CAST (jitterbuffer), percent);
|
|
gst_message_set_buffering_stats (message, GST_BUFFERING_LIVE, -1, -1, -1);
|
|
}
|
|
|
|
return message;
|
|
}
|
|
|
|
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 %d for seqnum %d to %" GST_TIME_FORMAT ", delay %"
|
|
GST_TIME_FORMAT, type, 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 (seqchange)
|
|
timer->num_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);
|
|
}
|
|
|
|
/* get the extra delay to wait before sending RTX */
|
|
static GstClockTime
|
|
get_rtx_delay (GstRtpJitterBufferPrivate * priv)
|
|
{
|
|
GstClockTime delay;
|
|
|
|
if (priv->rtx_delay == -1) {
|
|
if (priv->avg_jitter == 0)
|
|
delay = DEFAULT_AUTO_RTX_DELAY;
|
|
else
|
|
/* jitter is in nanoseconds, 2x jitter is a good margin */
|
|
delay = priv->avg_jitter * 2;
|
|
} else {
|
|
delay = priv->rtx_delay * GST_MSECOND;
|
|
}
|
|
if (priv->rtx_min_delay > 0)
|
|
delay = MAX (delay, priv->rtx_min_delay * GST_MSECOND);
|
|
|
|
return delay;
|
|
}
|
|
|
|
/* we just received a packet with seqnum and dts.
|
|
*
|
|
* First check for old seqnum that we are still expecting. If the gap with the
|
|
* current seqnum is too big, unschedule the timeouts.
|
|
*
|
|
* If we have a valid packet spacing estimate we can set a timer for when we
|
|
* should receive the next packet.
|
|
* If we don't have a valid estimate, we remove any timer we might have
|
|
* had for this packet.
|
|
*/
|
|
static void
|
|
update_timers (GstRtpJitterBuffer * jitterbuffer, guint16 seqnum,
|
|
GstClockTime dts, 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<->#%d gap %d", i,
|
|
test->type, test->seqnum, seqnum, gap);
|
|
|
|
if (gap == 0) {
|
|
GST_DEBUG ("found timer for current seqnum");
|
|
/* the timer for the current seqnum */
|
|
timer = test;
|
|
/* when no retransmission, we can stop now, we only need to find the
|
|
* timer for the current seqnum */
|
|
if (!priv->do_retransmission)
|
|
break;
|
|
} 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);
|
|
}
|
|
}
|
|
|
|
do_next_seqnum = do_next_seqnum && priv->packet_spacing > 0
|
|
&& priv->do_retransmission;
|
|
|
|
if (timer && timer->type != TIMER_TYPE_DEADLINE) {
|
|
if (timer->num_rtx_retry > 0) {
|
|
GstClockTime rtx_last, delay;
|
|
|
|
/* 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 != GST_CLOCK_TIME_NONE && dts > rtx_last) {
|
|
/* 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;
|
|
} else
|
|
delay = 0;
|
|
|
|
GST_LOG_OBJECT (jitterbuffer,
|
|
"RTX success %" G_GUINT64_FORMAT ", failed %" G_GUINT64_FORMAT
|
|
", requests %" G_GUINT64_FORMAT ", dups %" G_GUINT64_FORMAT
|
|
", avg-num %g, delay %" GST_TIME_FORMAT ", avg-rtt %" GST_TIME_FORMAT,
|
|
priv->num_rtx_success, priv->num_rtx_failed, priv->num_rtx_requests,
|
|
priv->num_duplicates, priv->avg_rtx_num, GST_TIME_ARGS (delay),
|
|
GST_TIME_ARGS (priv->avg_rtx_rtt));
|
|
|
|
/* don't try to estimate the next seqnum because this is a retransmitted
|
|
* packet and it probably did not arrive with the expected packet
|
|
* spacing. */
|
|
do_next_seqnum = FALSE;
|
|
}
|
|
}
|
|
|
|
if (do_next_seqnum) {
|
|
GstClockTime expected, delay;
|
|
|
|
/* calculate expected arrival time of the next seqnum */
|
|
expected = dts + priv->packet_spacing;
|
|
|
|
delay = get_rtx_delay (priv);
|
|
|
|
/* 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 we had a timer, remove it, we don't know when to expect the next
|
|
* packet. */
|
|
remove_timer (jitterbuffer, timer);
|
|
}
|
|
}
|
|
|
|
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 (gst_rtp_buffer_compare_seqnum (expected, seqnum) > 0) {
|
|
add_timer (jitterbuffer, type, expected, 0, expected_dts, 0, duration);
|
|
expected_dts += duration;
|
|
expected++;
|
|
}
|
|
}
|
|
|
|
static void
|
|
calculate_jitter (GstRtpJitterBuffer * jitterbuffer, GstClockTime dts,
|
|
guint rtptime)
|
|
{
|
|
gint32 rtpdiff;
|
|
GstClockTimeDiff dtsdiff, rtpdiffns, diff;
|
|
GstRtpJitterBufferPrivate *priv;
|
|
|
|
priv = jitterbuffer->priv;
|
|
|
|
if (G_UNLIKELY (dts == GST_CLOCK_TIME_NONE) || priv->clock_rate <= 0)
|
|
goto no_time;
|
|
|
|
if (priv->last_dts != -1)
|
|
dtsdiff = dts - priv->last_dts;
|
|
else
|
|
dtsdiff = 0;
|
|
|
|
if (priv->last_rtptime != -1)
|
|
rtpdiff = rtptime - (guint32) priv->last_rtptime;
|
|
else
|
|
rtpdiff = 0;
|
|
|
|
priv->last_dts = dts;
|
|
priv->last_rtptime = rtptime;
|
|
|
|
if (rtpdiff > 0)
|
|
rtpdiffns =
|
|
gst_util_uint64_scale_int (rtpdiff, GST_SECOND, priv->clock_rate);
|
|
else
|
|
rtpdiffns =
|
|
-gst_util_uint64_scale_int (-rtpdiff, GST_SECOND, priv->clock_rate);
|
|
|
|
diff = ABS (dtsdiff - rtpdiffns);
|
|
|
|
/* jitter is stored in nanoseconds */
|
|
priv->avg_jitter = (diff + (15 * priv->avg_jitter)) >> 4;
|
|
|
|
GST_LOG_OBJECT (jitterbuffer,
|
|
"dtsdiff %" GST_TIME_FORMAT " rtptime %" GST_TIME_FORMAT
|
|
", clock-rate %d, diff %" GST_TIME_FORMAT ", jitter: %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (dtsdiff), GST_TIME_ARGS (rtpdiffns), priv->clock_rate,
|
|
GST_TIME_ARGS (diff), GST_TIME_ARGS (priv->avg_jitter));
|
|
|
|
return;
|
|
|
|
/* ERRORS */
|
|
no_time:
|
|
{
|
|
GST_DEBUG_OBJECT (jitterbuffer,
|
|
"no dts or no clock-rate, can't calculate jitter");
|
|
return;
|
|
}
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_rtp_jitter_buffer_chain (GstPad * pad, GstObject * parent,
|
|
GstBuffer * buffer)
|
|
{
|
|
GstRtpJitterBuffer *jitterbuffer;
|
|
GstRtpJitterBufferPrivate *priv;
|
|
guint16 seqnum;
|
|
guint32 expected, rtptime;
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
GstClockTime dts, pts;
|
|
guint64 latency_ts;
|
|
gboolean head;
|
|
gint percent = -1;
|
|
guint8 pt;
|
|
GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
|
|
gboolean do_next_seqnum = FALSE;
|
|
RTPJitterBufferItem *item;
|
|
GstMessage *msg = NULL;
|
|
|
|
jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
|
|
|
|
priv = jitterbuffer->priv;
|
|
|
|
if (G_UNLIKELY (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp)))
|
|
goto invalid_buffer;
|
|
|
|
pt = gst_rtp_buffer_get_payload_type (&rtp);
|
|
seqnum = gst_rtp_buffer_get_seq (&rtp);
|
|
rtptime = gst_rtp_buffer_get_timestamp (&rtp);
|
|
gst_rtp_buffer_unmap (&rtp);
|
|
|
|
/* 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;
|
|
|
|
calculate_jitter (jitterbuffer, dts, rtptime);
|
|
|
|
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 (!GST_CLOCK_TIME_IS_VALID (dts)) {
|
|
/* We would run into calculations with GST_CLOCK_TIME_NONE below
|
|
* and can't compensate for anything without DTS on RTP packets
|
|
*/
|
|
goto gap_but_no_dts;
|
|
} else 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;
|
|
JBUF_SIGNAL_EVENT (priv);
|
|
}
|
|
/* 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_peek (priv->jbuf);
|
|
|
|
if (IS_DROPABLE (old_item)) {
|
|
old_item = rtp_jitter_buffer_pop (priv->jbuf, &percent);
|
|
GST_DEBUG_OBJECT (jitterbuffer, "Queue full, dropping old packet %p",
|
|
old_item);
|
|
priv->next_seqnum = (old_item->seqnum + 1) & 0xffff;
|
|
free_item (old_item);
|
|
}
|
|
/* we might have removed some head buffers, signal the pushing thread to
|
|
* see if it can push now */
|
|
JBUF_SIGNAL_EVENT (priv);
|
|
}
|
|
}
|
|
|
|
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,
|
|
&head, &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);
|
|
|
|
if (G_UNLIKELY (head)) {
|
|
/* signal addition of new buffer when the _loop is waiting. */
|
|
if (G_LIKELY (priv->active))
|
|
JBUF_SIGNAL_EVENT (priv);
|
|
|
|
/* let's unschedule and unblock any waiting buffers. We only want to do this
|
|
* when the head buffer changed */
|
|
if (G_UNLIKELY (priv->clock_id)) {
|
|
GST_DEBUG_OBJECT (jitterbuffer, "Unscheduling waiting new buffer");
|
|
unschedule_current_timer (jitterbuffer);
|
|
}
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer,
|
|
"Pushed packet #%d, now %d packets, head: %d, " "percent %d", seqnum,
|
|
rtp_jitter_buffer_num_packets (priv->jbuf), head, percent);
|
|
|
|
msg = check_buffering_percent (jitterbuffer, percent);
|
|
|
|
finished:
|
|
JBUF_UNLOCK (priv);
|
|
|
|
if (msg)
|
|
gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), msg);
|
|
|
|
return ret;
|
|
|
|
/* ERRORS */
|
|
invalid_buffer:
|
|
{
|
|
/* this is not fatal but should be filtered earlier */
|
|
GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
|
|
("Received invalid RTP payload, dropping"));
|
|
gst_buffer_unref (buffer);
|
|
return GST_FLOW_OK;
|
|
}
|
|
no_clock_rate:
|
|
{
|
|
GST_WARNING_OBJECT (jitterbuffer,
|
|
"No clock-rate in caps!, dropping buffer");
|
|
gst_buffer_unref (buffer);
|
|
goto finished;
|
|
}
|
|
out_flushing:
|
|
{
|
|
ret = priv->srcresult;
|
|
GST_DEBUG_OBJECT (jitterbuffer, "flushing %s", gst_flow_get_name (ret));
|
|
gst_buffer_unref (buffer);
|
|
goto finished;
|
|
}
|
|
have_eos:
|
|
{
|
|
ret = GST_FLOW_EOS;
|
|
GST_WARNING_OBJECT (jitterbuffer, "we are EOS, refusing buffer");
|
|
gst_buffer_unref (buffer);
|
|
goto finished;
|
|
}
|
|
too_late:
|
|
{
|
|
GST_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;
|
|
}
|
|
gap_but_no_dts:
|
|
{
|
|
/* this is fatal as we can't compensate for gaps without DTS */
|
|
GST_ELEMENT_ERROR (jitterbuffer, STREAM, DECODE, (NULL),
|
|
("Received packet without DTS after a gap"));
|
|
gst_buffer_unref (buffer);
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
}
|
|
|
|
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)
|
|
{
|
|
guint64 total, elapsed, left, estimated;
|
|
GstClockTime out_time;
|
|
GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
|
|
|
|
if (priv->npt_stop == -1 || priv->ext_timestamp == -1
|
|
|| priv->clock_base == -1 || priv->clock_rate <= 0)
|
|
return;
|
|
|
|
/* compute the elapsed time */
|
|
elapsed = compute_elapsed (jitterbuffer, item);
|
|
|
|
/* do nothing if elapsed time doesn't increment */
|
|
if (priv->last_elapsed && elapsed <= priv->last_elapsed)
|
|
return;
|
|
|
|
priv->last_elapsed = elapsed;
|
|
|
|
/* this is the total time we need to play */
|
|
total = priv->npt_stop - priv->npt_start;
|
|
GST_LOG_OBJECT (jitterbuffer, "total %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (total));
|
|
|
|
/* this is how much time there is left */
|
|
if (total > elapsed)
|
|
left = total - elapsed;
|
|
else
|
|
left = 0;
|
|
|
|
/* if we have less time left that the size of the buffer, we will not
|
|
* be able to keep it filled, disabled buffering then */
|
|
if (left < rtp_jitter_buffer_get_delay (priv->jbuf)) {
|
|
GST_DEBUG_OBJECT (jitterbuffer, "left %" GST_TIME_FORMAT
|
|
", disable buffering close to EOS", GST_TIME_ARGS (left));
|
|
rtp_jitter_buffer_disable_buffering (priv->jbuf, TRUE);
|
|
}
|
|
|
|
/* this is the current time as running-time */
|
|
out_time = item->dts;
|
|
|
|
if (elapsed > 0)
|
|
estimated = gst_util_uint64_scale (out_time, total, elapsed);
|
|
else {
|
|
/* if there is almost nothing left,
|
|
* we may never advance enough to end up in the above case */
|
|
if (total < GST_SECOND)
|
|
estimated = GST_SECOND;
|
|
else
|
|
estimated = -1;
|
|
}
|
|
GST_LOG_OBJECT (jitterbuffer, "elapsed %" GST_TIME_FORMAT ", estimated %"
|
|
GST_TIME_FORMAT, GST_TIME_ARGS (elapsed), GST_TIME_ARGS (estimated));
|
|
|
|
if (estimated != -1 && priv->estimated_eos != estimated) {
|
|
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 = GST_FLOW_OK;
|
|
RTPJitterBufferItem *item;
|
|
GstBuffer *outbuf = NULL;
|
|
GstEvent *outevent = NULL;
|
|
GstQuery *outquery = NULL;
|
|
GstClockTime dts, pts;
|
|
gint percent = -1;
|
|
gboolean do_push = TRUE;
|
|
guint type;
|
|
GstMessage *msg;
|
|
|
|
/* when we get here we are ready to pop and push the buffer */
|
|
item = rtp_jitter_buffer_pop (priv->jbuf, &percent);
|
|
type = item->type;
|
|
|
|
switch (type) {
|
|
case ITEM_TYPE_BUFFER:
|
|
|
|
/* we need to make writable to change the flags and timestamps */
|
|
outbuf = gst_buffer_make_writable (item->data);
|
|
|
|
if (G_UNLIKELY (priv->discont)) {
|
|
/* set DISCONT flag when we missed a packet. We pushed the buffer writable
|
|
* into the jitterbuffer so we can modify now. */
|
|
GST_DEBUG_OBJECT (jitterbuffer, "mark output buffer discont");
|
|
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
|
|
priv->discont = FALSE;
|
|
}
|
|
if (G_UNLIKELY (priv->ts_discont)) {
|
|
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
|
|
priv->ts_discont = FALSE;
|
|
}
|
|
|
|
dts =
|
|
gst_segment_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);
|
|
break;
|
|
case ITEM_TYPE_LOST:
|
|
priv->discont = TRUE;
|
|
if (!priv->do_lost)
|
|
do_push = FALSE;
|
|
/* FALLTHROUGH */
|
|
case ITEM_TYPE_EVENT:
|
|
outevent = item->data;
|
|
break;
|
|
case ITEM_TYPE_QUERY:
|
|
outquery = item->data;
|
|
break;
|
|
}
|
|
|
|
/* now we are ready to push the buffer. Save the seqnum and release the lock
|
|
* so the other end can push stuff in the queue again. */
|
|
if (seqnum != -1) {
|
|
priv->last_popped_seqnum = seqnum;
|
|
priv->next_seqnum = (seqnum + item->count) & 0xffff;
|
|
}
|
|
msg = check_buffering_percent (jitterbuffer, percent);
|
|
JBUF_UNLOCK (priv);
|
|
|
|
item->data = NULL;
|
|
free_item (item);
|
|
|
|
if (msg)
|
|
gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), msg);
|
|
|
|
switch (type) {
|
|
case ITEM_TYPE_BUFFER:
|
|
/* push buffer */
|
|
GST_DEBUG_OBJECT (jitterbuffer,
|
|
"Pushing buffer %d, dts %" GST_TIME_FORMAT ", pts %" GST_TIME_FORMAT,
|
|
seqnum, GST_TIME_ARGS (GST_BUFFER_DTS (outbuf)),
|
|
GST_TIME_ARGS (GST_BUFFER_PTS (outbuf)));
|
|
result = gst_pad_push (priv->srcpad, outbuf);
|
|
|
|
JBUF_LOCK_CHECK (priv, out_flushing);
|
|
break;
|
|
case ITEM_TYPE_LOST:
|
|
case ITEM_TYPE_EVENT:
|
|
GST_DEBUG_OBJECT (jitterbuffer, "%sPushing event %" GST_PTR_FORMAT
|
|
", seqnum %d", do_push ? "" : "NOT ", outevent, seqnum);
|
|
|
|
if (do_push)
|
|
gst_pad_push_event (priv->srcpad, outevent);
|
|
else
|
|
gst_event_unref (outevent);
|
|
|
|
result = GST_FLOW_OK;
|
|
|
|
JBUF_LOCK_CHECK (priv, out_flushing);
|
|
break;
|
|
case ITEM_TYPE_QUERY:
|
|
{
|
|
gboolean res;
|
|
|
|
res = gst_pad_peer_query (priv->srcpad, outquery);
|
|
|
|
JBUF_LOCK_CHECK (priv, out_flushing);
|
|
result = GST_FLOW_OK;
|
|
GST_LOG_OBJECT (jitterbuffer, "did query %p, return %d", outquery, res);
|
|
JBUF_SIGNAL_QUERY (priv, res);
|
|
break;
|
|
}
|
|
}
|
|
return result;
|
|
|
|
/* ERRORS */
|
|
out_flushing:
|
|
{
|
|
return priv->srcresult;
|
|
}
|
|
}
|
|
|
|
#define GST_FLOW_WAIT GST_FLOW_CUSTOM_SUCCESS
|
|
|
|
/* Peek a buffer and compare the seqnum to the expected seqnum.
|
|
* If all is fine, the buffer is pushed.
|
|
* If something is wrong, we wait for some event
|
|
*/
|
|
static GstFlowReturn
|
|
handle_next_buffer (GstRtpJitterBuffer * jitterbuffer)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
|
|
GstFlowReturn result = 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;
|
|
}
|
|
}
|
|
|
|
static GstClockTime
|
|
get_rtx_retry_timeout (GstRtpJitterBufferPrivate * priv)
|
|
{
|
|
GstClockTime rtx_retry_timeout;
|
|
GstClockTime rtx_min_retry_timeout;
|
|
|
|
if (priv->rtx_retry_timeout == -1) {
|
|
if (priv->avg_rtx_rtt == 0)
|
|
rtx_retry_timeout = DEFAULT_AUTO_RTX_TIMEOUT;
|
|
else
|
|
/* we want to ask for a retransmission after we waited for a
|
|
* complete RTT and the additional jitter */
|
|
rtx_retry_timeout = priv->avg_rtx_rtt + priv->avg_jitter * 2;
|
|
} else {
|
|
rtx_retry_timeout = priv->rtx_retry_timeout * GST_MSECOND;
|
|
}
|
|
/* make sure we don't retry too often. On very low latency networks,
|
|
* the RTT and jitter can be very low. */
|
|
if (priv->rtx_min_retry_timeout == -1) {
|
|
rtx_min_retry_timeout = priv->packet_spacing;
|
|
} else {
|
|
rtx_min_retry_timeout = priv->rtx_min_retry_timeout * GST_MSECOND;
|
|
}
|
|
rtx_retry_timeout = MAX (rtx_retry_timeout, rtx_min_retry_timeout);
|
|
|
|
return rtx_retry_timeout;
|
|
}
|
|
|
|
static GstClockTime
|
|
get_rtx_retry_period (GstRtpJitterBufferPrivate * priv,
|
|
GstClockTime rtx_retry_timeout)
|
|
{
|
|
GstClockTime rtx_retry_period;
|
|
|
|
if (priv->rtx_retry_period == -1) {
|
|
/* we retry up to the configured jitterbuffer size but leaving some
|
|
* room for the retransmission to arrive in time */
|
|
if (rtx_retry_timeout > priv->latency_ns) {
|
|
rtx_retry_period = 0;
|
|
} else {
|
|
rtx_retry_period = priv->latency_ns - rtx_retry_timeout;
|
|
}
|
|
} else {
|
|
rtx_retry_period = priv->rtx_retry_period * GST_MSECOND;
|
|
}
|
|
return rtx_retry_period;
|
|
}
|
|
|
|
/* 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, delay_ms, avg_rtx_rtt_ms;
|
|
guint rtx_retry_timeout_ms, rtx_retry_period_ms;
|
|
GstClockTime rtx_retry_period;
|
|
GstClockTime rtx_retry_timeout;
|
|
GstClock *clock;
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "expected %d didn't arrive, now %"
|
|
GST_TIME_FORMAT, timer->seqnum, GST_TIME_ARGS (now));
|
|
|
|
rtx_retry_timeout = get_rtx_retry_timeout (priv);
|
|
rtx_retry_period = get_rtx_retry_period (priv, rtx_retry_timeout);
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "timeout %" GST_TIME_FORMAT ", period %"
|
|
GST_TIME_FORMAT, GST_TIME_ARGS (rtx_retry_timeout),
|
|
GST_TIME_ARGS (rtx_retry_period));
|
|
|
|
delay = timer->rtx_delay + timer->rtx_retry;
|
|
|
|
delay_ms = GST_TIME_AS_MSECONDS (delay);
|
|
rtx_retry_timeout_ms = GST_TIME_AS_MSECONDS (rtx_retry_timeout);
|
|
rtx_retry_period_ms = GST_TIME_AS_MSECONDS (rtx_retry_period);
|
|
avg_rtx_rtt_ms = GST_TIME_AS_MSECONDS (priv->avg_rtx_rtt);
|
|
|
|
event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
|
|
gst_structure_new ("GstRTPRetransmissionRequest",
|
|
"seqnum", G_TYPE_UINT, (guint) timer->seqnum,
|
|
"running-time", G_TYPE_UINT64, timer->rtx_base,
|
|
"delay", G_TYPE_UINT, delay_ms,
|
|
"retry", G_TYPE_UINT, timer->num_rtx_retry,
|
|
"frequency", G_TYPE_UINT, rtx_retry_timeout_ms,
|
|
"period", G_TYPE_UINT, rtx_retry_period_ms,
|
|
"deadline", G_TYPE_UINT, priv->latency_ms,
|
|
"packet-spacing", G_TYPE_UINT64, priv->packet_spacing,
|
|
"avg-rtt", G_TYPE_UINT, avg_rtx_rtt_ms, NULL));
|
|
|
|
priv->num_rtx_requests++;
|
|
timer->num_rtx_retry++;
|
|
|
|
GST_OBJECT_LOCK (jitterbuffer);
|
|
if ((clock = GST_ELEMENT_CLOCK (jitterbuffer))) {
|
|
timer->rtx_last = gst_clock_get_time (clock);
|
|
timer->rtx_last -= GST_ELEMENT_CAST (jitterbuffer)->base_time;
|
|
} else {
|
|
timer->rtx_last = now;
|
|
}
|
|
GST_OBJECT_UNLOCK (jitterbuffer);
|
|
|
|
/* calculate the timeout for the next retransmission attempt */
|
|
timer->rtx_retry += rtx_retry_timeout;
|
|
GST_DEBUG_OBJECT (jitterbuffer, "base %" GST_TIME_FORMAT ", delay %"
|
|
GST_TIME_FORMAT ", retry %" GST_TIME_FORMAT ", num_retry %u",
|
|
GST_TIME_ARGS (timer->rtx_base), GST_TIME_ARGS (timer->rtx_delay),
|
|
GST_TIME_ARGS (timer->rtx_retry), timer->num_rtx_retry);
|
|
|
|
if (timer->rtx_retry + timer->rtx_delay > rtx_retry_period) {
|
|
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, next_in_seqnum;
|
|
gboolean late, head;
|
|
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;
|
|
|
|
next_in_seqnum = (seqnum + lost_packets) & 0xffff;
|
|
|
|
/* we now only accept seqnum bigger than this */
|
|
if (gst_rtp_buffer_compare_seqnum (priv->next_in_seqnum, next_in_seqnum) > 0)
|
|
priv->next_in_seqnum = next_in_seqnum;
|
|
|
|
/* 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, &head, NULL);
|
|
|
|
/* remove timer now */
|
|
remove_timer (jitterbuffer, timer);
|
|
if (head)
|
|
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);
|
|
if (!priv->eos) {
|
|
/* there was no EOS in the buffer, put one in there now */
|
|
queue_event (jitterbuffer, gst_event_new_eos ());
|
|
}
|
|
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");
|
|
|
|
/* timer seqnum might have been obsoleted by caps seqnum-base,
|
|
* only mess with current ongoing seqnum if still unknown */
|
|
if (priv->next_seqnum == -1)
|
|
priv->next_seqnum = timer->seqnum;
|
|
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
|
|
&& (gst_rtp_buffer_compare_seqnum (test->seqnum,
|
|
timer->seqnum) > 0))
|
|
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
|
|
&& (gst_rtp_buffer_compare_seqnum (test->seqnum,
|
|
timer->seqnum) > 0)) {
|
|
/* 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) {
|
|
gst_clock_id_unref (id);
|
|
priv->clock_id = NULL;
|
|
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 = GST_FLOW_OK;
|
|
|
|
priv = jitterbuffer->priv;
|
|
|
|
JBUF_LOCK_CHECK (priv, flushing);
|
|
do {
|
|
result = handle_next_buffer (jitterbuffer);
|
|
if (G_LIKELY (result == GST_FLOW_WAIT)) {
|
|
/* now wait for the next event */
|
|
JBUF_WAIT_EVENT (priv, flushing);
|
|
result = GST_FLOW_OK;
|
|
}
|
|
}
|
|
while (result == GST_FLOW_OK);
|
|
/* store result for upstream */
|
|
priv->srcresult = result;
|
|
/* if we get here we need to pause */
|
|
goto pause;
|
|
|
|
/* ERRORS */
|
|
flushing:
|
|
{
|
|
result = priv->srcresult;
|
|
goto pause;
|
|
}
|
|
pause:
|
|
{
|
|
GstEvent *event;
|
|
|
|
JBUF_SIGNAL_QUERY (priv, FALSE);
|
|
JBUF_UNLOCK (priv);
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "pausing task, reason %s",
|
|
gst_flow_get_name (result));
|
|
gst_pad_pause_task (priv->srcpad);
|
|
if (result == GST_FLOW_EOS) {
|
|
event = gst_event_new_eos ();
|
|
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 valid = TRUE, keep = FALSE;
|
|
|
|
priv = jitterbuffer->priv;
|
|
|
|
/* get the last values from the jitterbuffer */
|
|
rtp_jitter_buffer_get_sync (priv->jbuf, &base_rtptime, &base_time,
|
|
&clock_rate, &last_rtptime);
|
|
|
|
clock_base = priv->clock_base;
|
|
ext_rtptime = priv->ext_rtptime;
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "ext SR %" G_GUINT64_FORMAT ", base %"
|
|
G_GUINT64_FORMAT ", clock-rate %" G_GUINT32_FORMAT
|
|
", clock-base %" G_GUINT64_FORMAT ", last-rtptime %" G_GUINT64_FORMAT,
|
|
ext_rtptime, base_rtptime, clock_rate, clock_base, last_rtptime);
|
|
|
|
if (base_rtptime == -1 || clock_rate == -1 || base_time == -1) {
|
|
/* we keep this SR packet for later. When we get a valid RTP packet the
|
|
* above values will be set and we can try to use the SR packet */
|
|
GST_DEBUG_OBJECT (jitterbuffer, "keeping for later, no RTP values");
|
|
keep = TRUE;
|
|
} else {
|
|
/* we can't accept anything that happened before we did the last resync */
|
|
if (base_rtptime > ext_rtptime) {
|
|
GST_DEBUG_OBJECT (jitterbuffer, "dropping, older than base time");
|
|
valid = FALSE;
|
|
} else {
|
|
/* the SR RTP timestamp must be something close to what we last observed
|
|
* in the jitterbuffer */
|
|
if (ext_rtptime > last_rtptime) {
|
|
/* check how far ahead it is to our RTP timestamps */
|
|
diff = ext_rtptime - last_rtptime;
|
|
/* if bigger than 1 second, we drop it */
|
|
if (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 (keep) {
|
|
GST_DEBUG_OBJECT (jitterbuffer, "keeping RTCP packet for later");
|
|
} else if (valid) {
|
|
GstStructure *s;
|
|
|
|
s = gst_structure_new ("application/x-rtp-sync",
|
|
"base-rtptime", G_TYPE_UINT64, base_rtptime,
|
|
"base-time", G_TYPE_UINT64, base_time,
|
|
"clock-rate", G_TYPE_UINT, clock_rate,
|
|
"clock-base", G_TYPE_UINT64, clock_base,
|
|
"sr-ext-rtptime", G_TYPE_UINT64, ext_rtptime,
|
|
"sr-buffer", GST_TYPE_BUFFER, priv->last_sr, NULL);
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "signaling sync");
|
|
gst_buffer_replace (&priv->last_sr, NULL);
|
|
JBUF_UNLOCK (priv);
|
|
g_signal_emit (jitterbuffer,
|
|
gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC], 0, s);
|
|
JBUF_LOCK (priv);
|
|
gst_structure_free (s);
|
|
} else {
|
|
GST_DEBUG_OBJECT (jitterbuffer, "dropping RTCP packet");
|
|
gst_buffer_replace (&priv->last_sr, NULL);
|
|
}
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad, GstObject * parent,
|
|
GstBuffer * buffer)
|
|
{
|
|
GstRtpJitterBuffer *jitterbuffer;
|
|
GstRtpJitterBufferPrivate *priv;
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
guint32 ssrc;
|
|
GstRTCPPacket packet;
|
|
guint64 ext_rtptime;
|
|
guint32 rtptime;
|
|
GstRTCPBuffer rtcp = { NULL, };
|
|
|
|
jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
|
|
|
|
if (G_UNLIKELY (!gst_rtcp_buffer_validate (buffer)))
|
|
goto invalid_buffer;
|
|
|
|
priv = jitterbuffer->priv;
|
|
|
|
gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
|
|
|
|
if (!gst_rtcp_buffer_get_first_packet (&rtcp, &packet))
|
|
goto empty_buffer;
|
|
|
|
/* first packet must be SR or RR or else the validate would have failed */
|
|
switch (gst_rtcp_packet_get_type (&packet)) {
|
|
case GST_RTCP_TYPE_SR:
|
|
gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, NULL, &rtptime,
|
|
NULL, NULL);
|
|
break;
|
|
default:
|
|
goto ignore_buffer;
|
|
}
|
|
gst_rtcp_buffer_unmap (&rtcp);
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "received RTCP of SSRC %08x", ssrc);
|
|
|
|
JBUF_LOCK (priv);
|
|
/* convert the RTP timestamp to our extended timestamp, using the same offset
|
|
* we used in the jitterbuffer */
|
|
ext_rtptime = priv->jbuf->ext_rtptime;
|
|
ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
|
|
|
|
priv->ext_rtptime = ext_rtptime;
|
|
gst_buffer_replace (&priv->last_sr, buffer);
|
|
|
|
do_handle_sync (jitterbuffer);
|
|
JBUF_UNLOCK (priv);
|
|
|
|
done:
|
|
gst_buffer_unref (buffer);
|
|
|
|
return ret;
|
|
|
|
invalid_buffer:
|
|
{
|
|
/* this is not fatal but should be filtered earlier */
|
|
GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
|
|
("Received invalid RTCP payload, dropping"));
|
|
ret = GST_FLOW_OK;
|
|
goto done;
|
|
}
|
|
empty_buffer:
|
|
{
|
|
/* this is not fatal but should be filtered earlier */
|
|
GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
|
|
("Received empty RTCP payload, dropping"));
|
|
gst_rtcp_buffer_unmap (&rtcp);
|
|
ret = GST_FLOW_OK;
|
|
goto done;
|
|
}
|
|
ignore_buffer:
|
|
{
|
|
GST_DEBUG_OBJECT (jitterbuffer, "ignoring RTCP packet");
|
|
gst_rtcp_buffer_unmap (&rtcp);
|
|
ret = GST_FLOW_OK;
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_jitter_buffer_sink_query (GstPad * pad, GstObject * parent,
|
|
GstQuery * query)
|
|
{
|
|
gboolean res = FALSE;
|
|
GstRtpJitterBuffer *jitterbuffer;
|
|
GstRtpJitterBufferPrivate *priv;
|
|
|
|
jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
|
|
priv = jitterbuffer->priv;
|
|
|
|
switch (GST_QUERY_TYPE (query)) {
|
|
case GST_QUERY_CAPS:
|
|
{
|
|
GstCaps *filter, *caps;
|
|
|
|
gst_query_parse_caps (query, &filter);
|
|
caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
|
|
gst_query_set_caps_result (query, caps);
|
|
gst_caps_unref (caps);
|
|
res = TRUE;
|
|
break;
|
|
}
|
|
default:
|
|
if (GST_QUERY_IS_SERIALIZED (query)) {
|
|
RTPJitterBufferItem *item;
|
|
gboolean head;
|
|
|
|
JBUF_LOCK_CHECK (priv, out_flushing);
|
|
if (rtp_jitter_buffer_get_mode (priv->jbuf) !=
|
|
RTP_JITTER_BUFFER_MODE_BUFFER) {
|
|
GST_DEBUG_OBJECT (jitterbuffer, "adding serialized query");
|
|
item = alloc_item (query, ITEM_TYPE_QUERY, -1, -1, -1, 0, -1);
|
|
rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
|
|
if (head)
|
|
JBUF_SIGNAL_EVENT (priv);
|
|
JBUF_WAIT_QUERY (priv, out_flushing);
|
|
res = priv->last_query;
|
|
} else {
|
|
GST_DEBUG_OBJECT (jitterbuffer, "refusing query, we are buffering");
|
|
res = FALSE;
|
|
}
|
|
JBUF_UNLOCK (priv);
|
|
} else {
|
|
res = gst_pad_query_default (pad, parent, query);
|
|
}
|
|
break;
|
|
}
|
|
return res;
|
|
/* ERRORS */
|
|
out_flushing:
|
|
{
|
|
GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
|
|
JBUF_UNLOCK (priv);
|
|
return FALSE;
|
|
}
|
|
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_jitter_buffer_src_query (GstPad * pad, GstObject * parent,
|
|
GstQuery * query)
|
|
{
|
|
GstRtpJitterBuffer *jitterbuffer;
|
|
GstRtpJitterBufferPrivate *priv;
|
|
gboolean res = FALSE;
|
|
|
|
jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
|
|
priv = jitterbuffer->priv;
|
|
|
|
switch (GST_QUERY_TYPE (query)) {
|
|
case GST_QUERY_LATENCY:
|
|
{
|
|
/* We need to send the query upstream and add the returned latency to our
|
|
* own */
|
|
GstClockTime min_latency, max_latency;
|
|
gboolean us_live;
|
|
GstClockTime our_latency;
|
|
|
|
if ((res = gst_pad_peer_query (priv->sinkpad, query))) {
|
|
gst_query_parse_latency (query, &us_live, &min_latency, &max_latency);
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "Peer latency: min %"
|
|
GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
|
|
|
|
/* store this so that we can safely sync on the peer buffers. */
|
|
JBUF_LOCK (priv);
|
|
priv->peer_latency = min_latency;
|
|
our_latency = priv->latency_ns;
|
|
JBUF_UNLOCK (priv);
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "Our latency: %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (our_latency));
|
|
|
|
/* we add some latency but can buffer an infinite amount of time */
|
|
min_latency += our_latency;
|
|
max_latency = -1;
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "Calculated total latency : min %"
|
|
GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
|
|
|
|
gst_query_set_latency (query, TRUE, min_latency, max_latency);
|
|
}
|
|
break;
|
|
}
|
|
case GST_QUERY_POSITION:
|
|
{
|
|
GstClockTime start, last_out;
|
|
GstFormat fmt;
|
|
|
|
gst_query_parse_position (query, &fmt, NULL);
|
|
if (fmt != GST_FORMAT_TIME) {
|
|
res = gst_pad_query_default (pad, parent, query);
|
|
break;
|
|
}
|
|
|
|
JBUF_LOCK (priv);
|
|
start = priv->npt_start;
|
|
last_out = priv->last_out_time;
|
|
JBUF_UNLOCK (priv);
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "npt start %" GST_TIME_FORMAT
|
|
", last out %" GST_TIME_FORMAT, GST_TIME_ARGS (start),
|
|
GST_TIME_ARGS (last_out));
|
|
|
|
if (GST_CLOCK_TIME_IS_VALID (start) && GST_CLOCK_TIME_IS_VALID (last_out)) {
|
|
/* bring 0-based outgoing time to stream time */
|
|
gst_query_set_position (query, GST_FORMAT_TIME, start + last_out);
|
|
res = TRUE;
|
|
} else {
|
|
res = gst_pad_query_default (pad, parent, query);
|
|
}
|
|
break;
|
|
}
|
|
case GST_QUERY_CAPS:
|
|
{
|
|
GstCaps *filter, *caps;
|
|
|
|
gst_query_parse_caps (query, &filter);
|
|
caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
|
|
gst_query_set_caps_result (query, caps);
|
|
gst_caps_unref (caps);
|
|
res = TRUE;
|
|
break;
|
|
}
|
|
default:
|
|
res = gst_pad_query_default (pad, parent, query);
|
|
break;
|
|
}
|
|
|
|
return res;
|
|
}
|
|
|
|
static void
|
|
gst_rtp_jitter_buffer_set_property (GObject * object,
|
|
guint prop_id, const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstRtpJitterBuffer *jitterbuffer;
|
|
GstRtpJitterBufferPrivate *priv;
|
|
|
|
jitterbuffer = GST_RTP_JITTER_BUFFER (object);
|
|
priv = jitterbuffer->priv;
|
|
|
|
switch (prop_id) {
|
|
case PROP_LATENCY:
|
|
{
|
|
guint new_latency, old_latency;
|
|
|
|
new_latency = g_value_get_uint (value);
|
|
|
|
JBUF_LOCK (priv);
|
|
old_latency = priv->latency_ms;
|
|
priv->latency_ms = new_latency;
|
|
priv->latency_ns = priv->latency_ms * GST_MSECOND;
|
|
rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
|
|
JBUF_UNLOCK (priv);
|
|
|
|
/* post message if latency changed, this will inform the parent pipeline
|
|
* that a latency reconfiguration is possible/needed. */
|
|
if (new_latency != old_latency) {
|
|
GST_DEBUG_OBJECT (jitterbuffer, "latency changed to: %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (new_latency * GST_MSECOND));
|
|
|
|
gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer),
|
|
gst_message_new_latency (GST_OBJECT_CAST (jitterbuffer)));
|
|
}
|
|
break;
|
|
}
|
|
case PROP_DROP_ON_LATENCY:
|
|
JBUF_LOCK (priv);
|
|
priv->drop_on_latency = g_value_get_boolean (value);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_TS_OFFSET:
|
|
JBUF_LOCK (priv);
|
|
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_MIN_DELAY:
|
|
JBUF_LOCK (priv);
|
|
priv->rtx_min_delay = g_value_get_uint (value);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_RTX_DELAY_REORDER:
|
|
JBUF_LOCK (priv);
|
|
priv->rtx_delay_reorder = g_value_get_int (value);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_RTX_RETRY_TIMEOUT:
|
|
JBUF_LOCK (priv);
|
|
priv->rtx_retry_timeout = g_value_get_int (value);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_RTX_MIN_RETRY_TIMEOUT:
|
|
JBUF_LOCK (priv);
|
|
priv->rtx_min_retry_timeout = g_value_get_int (value);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_RTX_RETRY_PERIOD:
|
|
JBUF_LOCK (priv);
|
|
priv->rtx_retry_period = g_value_get_int (value);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
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_MIN_DELAY:
|
|
JBUF_LOCK (priv);
|
|
g_value_set_uint (value, priv->rtx_min_delay);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_RTX_DELAY_REORDER:
|
|
JBUF_LOCK (priv);
|
|
g_value_set_int (value, priv->rtx_delay_reorder);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_RTX_RETRY_TIMEOUT:
|
|
JBUF_LOCK (priv);
|
|
g_value_set_int (value, priv->rtx_retry_timeout);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_RTX_MIN_RETRY_TIMEOUT:
|
|
JBUF_LOCK (priv);
|
|
g_value_set_int (value, priv->rtx_min_retry_timeout);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_RTX_RETRY_PERIOD:
|
|
JBUF_LOCK (priv);
|
|
g_value_set_int (value, priv->rtx_retry_period);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_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;
|
|
}
|