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gstreamer/gst/rtpmanager/gstrtpjitterbuffer.c
Alessandro Decina 40899379c0 rtpjitterbuffer: move some initialization code from change_state to _init. 
Set ->active to TRUE in _init so it can be set to FALSE after creating the
jitterbuffer and it won't be mistakenly reset to TRUE in the change_state
function.
This is needed to start the jitterbuffer as inactive when rtpbin is buffering.
2010-05-03 13:34:59 +02:00

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/*
* Farsight Voice+Video library
*
* Copyright 2007 Collabora Ltd,
* Copyright 2007 Nokia Corporation
* @author: Philippe Kalaf <philippe.kalaf@collabora.co.uk>.
* Copyright 2007 Wim Taymans <wim.taymans@gmail.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*
*/
/**
* SECTION:element-gstrtpjitterbuffer
*
* This element reorders and removes duplicate RTP packets as they are received
* from a network source. It will also wait for missing packets up to a
* configurable time limit using the #GstRtpJitterBuffer:latency property.
* Packets arriving too late are considered to be lost packets.
*
* This element acts as a live element and so adds #GstRtpJitterBuffer:latency
* to the pipeline.
*
* The element needs the clock-rate of the RTP payload in order to estimate the
* delay. This information is obtained either from the caps on the sink pad or,
* when no caps are present, from the #GstRtpJitterBuffer::request-pt-map signal.
* To clear the previous pt-map use the #GstRtpJitterBuffer::clear-pt-map signal.
*
* This element will automatically be used inside gstrtpbin.
*
* <refsect2>
* <title>Example pipelines</title>
* |[
* gst-launch rtspsrc location=rtsp://192.168.1.133:8554/mpeg1or2AudioVideoTest ! gstrtpjitterbuffer ! rtpmpvdepay ! mpeg2dec ! xvimagesink
* ]| Connect to a streaming server and decode the MPEG video. The jitterbuffer is
* inserted into the pipeline to smooth out network jitter and to reorder the
* out-of-order RTP packets.
* </refsect2>
*
* Last reviewed on 2007-05-28 (0.10.5)
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <stdlib.h>
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include "gstrtpbin-marshal.h"
#include "gstrtpjitterbuffer.h"
#include "rtpjitterbuffer.h"
#include "rtpstats.h"
GST_DEBUG_CATEGORY (rtpjitterbuffer_debug);
#define GST_CAT_DEFAULT (rtpjitterbuffer_debug)
/* RTPJitterBuffer signals and args */
enum
{
SIGNAL_REQUEST_PT_MAP,
SIGNAL_CLEAR_PT_MAP,
SIGNAL_HANDLE_SYNC,
SIGNAL_ON_NPT_STOP,
SIGNAL_SET_ACTIVE,
LAST_SIGNAL
};
#define DEFAULT_LATENCY_MS 200
#define DEFAULT_DROP_ON_LATENCY FALSE
#define DEFAULT_TS_OFFSET 0
#define DEFAULT_DO_LOST FALSE
#define DEFAULT_MODE RTP_JITTER_BUFFER_MODE_SLAVE
#define DEFAULT_PERCENT 0
enum
{
PROP_0,
PROP_LATENCY,
PROP_DROP_ON_LATENCY,
PROP_TS_OFFSET,
PROP_DO_LOST,
PROP_MODE,
PROP_PERCENT,
PROP_LAST
};
#define JBUF_LOCK(priv) (g_mutex_lock ((priv)->jbuf_lock))
#define JBUF_LOCK_CHECK(priv,label) G_STMT_START { \
JBUF_LOCK (priv); \
if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
goto label; \
} G_STMT_END
#define JBUF_UNLOCK(priv) (g_mutex_unlock ((priv)->jbuf_lock))
#define JBUF_WAIT(priv) (g_cond_wait ((priv)->jbuf_cond, (priv)->jbuf_lock))
#define JBUF_WAIT_CHECK(priv,label) G_STMT_START { \
JBUF_WAIT(priv); \
if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
goto label; \
} G_STMT_END
#define JBUF_SIGNAL(priv) (g_cond_signal ((priv)->jbuf_cond))
struct _GstRtpJitterBufferPrivate
{
GstPad *sinkpad, *srcpad;
GstPad *rtcpsinkpad;
RTPJitterBuffer *jbuf;
GMutex *jbuf_lock;
GCond *jbuf_cond;
gboolean waiting;
gboolean discont;
gboolean active;
guint64 out_offset;
/* properties */
guint latency_ms;
guint64 latency_ns;
gboolean drop_on_latency;
gint64 ts_offset;
gboolean do_lost;
/* the last seqnum we pushed out */
guint32 last_popped_seqnum;
/* the next expected seqnum we push */
guint32 next_seqnum;
/* last output time */
GstClockTime last_out_time;
/* the next expected seqnum we receive */
guint32 next_in_seqnum;
/* start and stop ranges */
GstClockTime npt_start;
GstClockTime npt_stop;
guint64 ext_timestamp;
guint64 last_elapsed;
guint64 estimated_eos;
GstClockID eos_id;
gboolean reached_npt_stop;
/* state */
gboolean eos;
/* clock rate and rtp timestamp offset */
gint last_pt;
gint32 clock_rate;
gint64 clock_base;
gint64 prev_ts_offset;
/* when we are shutting down */
GstFlowReturn srcresult;
gboolean blocked;
/* for sync */
GstSegment segment;
GstClockID clock_id;
gboolean unscheduled;
/* the latency of the upstream peer, we have to take this into account when
* synchronizing the buffers. */
GstClockTime peer_latency;
/* some accounting */
guint64 num_late;
guint64 num_duplicates;
};
#define GST_RTP_JITTER_BUFFER_GET_PRIVATE(o) \
(G_TYPE_INSTANCE_GET_PRIVATE ((o), GST_TYPE_RTP_JITTER_BUFFER, \
GstRtpJitterBufferPrivate))
static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"clock-rate = (int) [ 1, 2147483647 ]"
/* "payload = (int) , "
* "encoding-name = (string) "
*/ )
);
static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_rtcp_template =
GST_STATIC_PAD_TEMPLATE ("sink_rtcp",
GST_PAD_SINK,
GST_PAD_REQUEST,
GST_STATIC_CAPS ("application/x-rtcp")
);
static GstStaticPadTemplate gst_rtp_jitter_buffer_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp"
/* "payload = (int) , "
* "clock-rate = (int) , "
* "encoding-name = (string) "
*/ )
);
static guint gst_rtp_jitter_buffer_signals[LAST_SIGNAL] = { 0 };
GST_BOILERPLATE (GstRtpJitterBuffer, gst_rtp_jitter_buffer, GstElement,
GST_TYPE_ELEMENT);
/* object overrides */
static void gst_rtp_jitter_buffer_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_rtp_jitter_buffer_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
static void gst_rtp_jitter_buffer_finalize (GObject * object);
/* element overrides */
static GstStateChangeReturn gst_rtp_jitter_buffer_change_state (GstElement
* element, GstStateChange transition);
static GstPad *gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
GstPadTemplate * templ, const gchar * name);
static void gst_rtp_jitter_buffer_release_pad (GstElement * element,
GstPad * pad);
/* pad overrides */
static GstCaps *gst_rtp_jitter_buffer_getcaps (GstPad * pad);
static GstIterator *gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad);
/* sinkpad overrides */
static gboolean gst_jitter_buffer_sink_setcaps (GstPad * pad, GstCaps * caps);
static gboolean gst_rtp_jitter_buffer_sink_event (GstPad * pad,
GstEvent * event);
static GstFlowReturn gst_rtp_jitter_buffer_chain (GstPad * pad,
GstBuffer * buffer);
static gboolean gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad,
GstEvent * event);
static GstFlowReturn gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad,
GstBuffer * buffer);
/* srcpad overrides */
static gboolean gst_rtp_jitter_buffer_src_event (GstPad * pad,
GstEvent * event);
static gboolean
gst_rtp_jitter_buffer_src_activate_push (GstPad * pad, gboolean active);
static void gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer);
static gboolean gst_rtp_jitter_buffer_query (GstPad * pad, GstQuery * query);
static void
gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer);
static GstClockTime
gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jitterbuffer,
gboolean active, guint64 base_time);
static void
gst_rtp_jitter_buffer_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_jitter_buffer_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_jitter_buffer_sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_jitter_buffer_sink_rtcp_template));
gst_element_class_set_details_simple (element_class,
"RTP packet jitter-buffer", "Filter/Network/RTP",
"A buffer that deals with network jitter and other transmission faults",
"Philippe Kalaf <philippe.kalaf@collabora.co.uk>, "
"Wim Taymans <wim.taymans@gmail.com>");
}
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_DEBUG_FUNCPTR (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));
/**
* 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));
/**
* GstRtpJitterBuffer::ts-offset:
*
* Adjust GStreamer output buffer timestamps in the jitterbuffer with offset.
* This is mainly used to ensure interstream synchronisation.
*/
g_object_class_install_property (gobject_class, PROP_TS_OFFSET,
g_param_spec_int64 ("ts-offset", "Timestamp Offset",
"Adjust buffer timestamps with offset in nanoseconds", G_MININT64,
G_MAXINT64, DEFAULT_TS_OFFSET,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRtpJitterBuffer::do-lost:
*
* Send out a GstRTPPacketLost event downstream when a packet is considered
* lost.
*/
g_object_class_install_property (gobject_class, PROP_DO_LOST,
g_param_spec_boolean ("do-lost", "Do Lost",
"Send an event downstream when a packet is lost", DEFAULT_DO_LOST,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRtpJitterBuffer::mode:
*
* Control the buffering and timestamping mode used by the jitterbuffer.
*/
g_object_class_install_property (gobject_class, PROP_MODE,
g_param_spec_enum ("mode", "Mode",
"Control the buffering algorithm in use", RTP_TYPE_JITTER_BUFFER_MODE,
DEFAULT_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRtpJitterBuffer::percent:
*
* The percent of the jitterbuffer that is filled.
*
* Since: 0.10.19
*/
g_object_class_install_property (gobject_class, PROP_PERCENT,
g_param_spec_int ("percent", "percent",
"The buffer filled percent", 0, 100,
0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
/**
* GstRtpJitterBuffer::request-pt-map:
* @buffer: the object which received the signal
* @pt: the pt
*
* Request the payload type as #GstCaps for @pt.
*/
gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP] =
g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
request_pt_map), NULL, NULL, gst_rtp_bin_marshal_BOXED__UINT,
GST_TYPE_CAPS, 1, G_TYPE_UINT);
/**
* GstRtpJitterBuffer::handle-sync:
* @buffer: the object which received the signal
* @struct: a GstStructure containing sync values.
*
* Be notified of new sync values.
*/
gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC] =
g_signal_new ("handle-sync", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
handle_sync), NULL, NULL, g_cclosure_marshal_VOID__BOXED,
G_TYPE_NONE, 1, GST_TYPE_STRUCTURE | G_SIGNAL_TYPE_STATIC_SCOPE);
/**
* GstRtpJitterBuffer::on-npt-stop
* @buffer: the object which received the signal
*
* Signal that the jitterbufer has pushed the RTP packet that corresponds to
* the npt-stop position.
*/
gst_rtp_jitter_buffer_signals[SIGNAL_ON_NPT_STOP] =
g_signal_new ("on-npt-stop", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
on_npt_stop), NULL, NULL, g_cclosure_marshal_VOID__VOID,
G_TYPE_NONE, 0, G_TYPE_NONE);
/**
* GstRtpJitterBuffer::clear-pt-map:
* @buffer: the object which received the signal
*
* Invalidate the clock-rate as obtained with the
* #GstRtpJitterBuffer::request-pt-map signal.
*/
gst_rtp_jitter_buffer_signals[SIGNAL_CLEAR_PT_MAP] =
g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
G_STRUCT_OFFSET (GstRtpJitterBufferClass, clear_pt_map), NULL, NULL,
g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
/**
* GstRtpJitterBuffer::set-active:
* @buffer: the object which received the signal
*
* Start pushing out packets with the given base time. This signal is only
* useful in buffering mode.
*
* Returns: the time of the last pushed packet.
*
* Since: 0.10.19
*/
gst_rtp_jitter_buffer_signals[SIGNAL_SET_ACTIVE] =
g_signal_new ("set-active", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
G_STRUCT_OFFSET (GstRtpJitterBufferClass, set_active), NULL, NULL,
gst_rtp_bin_marshal_UINT64__BOOL_UINT64, G_TYPE_UINT64, 2, G_TYPE_BOOLEAN,
G_TYPE_UINT64);
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_change_state);
gstelement_class->request_new_pad =
GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_request_new_pad);
gstelement_class->release_pad =
GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_release_pad);
klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_clear_pt_map);
klass->set_active = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_set_active);
GST_DEBUG_CATEGORY_INIT
(rtpjitterbuffer_debug, "gstrtpjitterbuffer", 0, "RTP Jitter Buffer");
}
static void
gst_rtp_jitter_buffer_init (GstRtpJitterBuffer * jitterbuffer,
GstRtpJitterBufferClass * klass)
{
GstRtpJitterBufferPrivate *priv;
priv = GST_RTP_JITTER_BUFFER_GET_PRIVATE (jitterbuffer);
jitterbuffer->priv = priv;
priv->latency_ms = DEFAULT_LATENCY_MS;
priv->latency_ns = priv->latency_ms * GST_MSECOND;
priv->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
priv->do_lost = DEFAULT_DO_LOST;
priv->jbuf = rtp_jitter_buffer_new ();
priv->jbuf_lock = g_mutex_new ();
priv->jbuf_cond = g_cond_new ();
/* reset skew detection initialy */
rtp_jitter_buffer_reset_skew (priv->jbuf);
rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
rtp_jitter_buffer_set_buffering (priv->jbuf, FALSE);
priv->active = TRUE;
priv->srcpad =
gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_src_template,
"src");
gst_pad_set_activatepush_function (priv->srcpad,
GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_activate_push));
gst_pad_set_query_function (priv->srcpad,
GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_query));
gst_pad_set_getcaps_function (priv->srcpad,
GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_getcaps));
gst_pad_set_event_function (priv->srcpad,
GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_event));
priv->sinkpad =
gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_sink_template,
"sink");
gst_pad_set_chain_function (priv->sinkpad,
GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_chain));
gst_pad_set_event_function (priv->sinkpad,
GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_event));
gst_pad_set_setcaps_function (priv->sinkpad,
GST_DEBUG_FUNCPTR (gst_jitter_buffer_sink_setcaps));
gst_pad_set_getcaps_function (priv->sinkpad,
GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_getcaps));
gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->srcpad);
gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->sinkpad);
}
static void
gst_rtp_jitter_buffer_finalize (GObject * object)
{
GstRtpJitterBuffer *jitterbuffer;
jitterbuffer = GST_RTP_JITTER_BUFFER (object);
g_mutex_free (jitterbuffer->priv->jbuf_lock);
g_cond_free (jitterbuffer->priv->jbuf_cond);
g_object_unref (jitterbuffer->priv->jbuf);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static GstIterator *
gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad)
{
GstRtpJitterBuffer *jitterbuffer;
GstPad *otherpad = NULL;
GstIterator *it;
jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
if (pad == jitterbuffer->priv->sinkpad) {
otherpad = jitterbuffer->priv->srcpad;
} else if (pad == jitterbuffer->priv->srcpad) {
otherpad = jitterbuffer->priv->sinkpad;
} else if (pad == jitterbuffer->priv->rtcpsinkpad) {
otherpad = NULL;
}
it = gst_iterator_new_single (GST_TYPE_PAD, otherpad,
(GstCopyFunction) gst_object_ref, (GFreeFunc) gst_object_unref);
gst_object_unref (jitterbuffer);
return it;
}
static GstPad *
create_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
{
GstRtpJitterBufferPrivate *priv;
priv = jitterbuffer->priv;
GST_DEBUG_OBJECT (jitterbuffer, "creating RTCP sink pad");
priv->rtcpsinkpad =
gst_pad_new_from_static_template
(&gst_rtp_jitter_buffer_sink_rtcp_template, "sink_rtcp");
gst_pad_set_chain_function (priv->rtcpsinkpad,
gst_rtp_jitter_buffer_chain_rtcp);
gst_pad_set_event_function (priv->rtcpsinkpad,
(GstPadEventFunction) gst_rtp_jitter_buffer_sink_rtcp_event);
gst_pad_set_iterate_internal_links_function (priv->rtcpsinkpad,
gst_rtp_jitter_buffer_iterate_internal_links);
gst_pad_set_active (priv->rtcpsinkpad, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
return priv->rtcpsinkpad;
}
static void
remove_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
{
GstRtpJitterBufferPrivate *priv;
priv = jitterbuffer->priv;
GST_DEBUG_OBJECT (jitterbuffer, "removing RTCP sink pad");
gst_pad_set_active (priv->rtcpsinkpad, FALSE);
gst_element_remove_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
priv->rtcpsinkpad = NULL;
}
static GstPad *
gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
GstPadTemplate * templ, const gchar * name)
{
GstRtpJitterBuffer *jitterbuffer;
GstElementClass *klass;
GstPad *result;
GstRtpJitterBufferPrivate *priv;
g_return_val_if_fail (templ != NULL, NULL);
g_return_val_if_fail (GST_IS_RTP_JITTER_BUFFER (element), NULL);
jitterbuffer = GST_RTP_JITTER_BUFFER (element);
priv = jitterbuffer->priv;
klass = GST_ELEMENT_GET_CLASS (element);
GST_DEBUG_OBJECT (element, "requesting pad %s", GST_STR_NULL (name));
/* figure out the template */
if (templ == gst_element_class_get_pad_template (klass, "sink_rtcp")) {
if (priv->rtcpsinkpad != NULL)
goto exists;
result = create_rtcp_sink (jitterbuffer);
} else
goto wrong_template;
return result;
/* ERRORS */
wrong_template:
{
g_warning ("gstrtpjitterbuffer: this is not our template");
return NULL;
}
exists:
{
g_warning ("gstrtpjitterbuffer: pad already requested");
return NULL;
}
}
static void
gst_rtp_jitter_buffer_release_pad (GstElement * element, GstPad * pad)
{
GstRtpJitterBuffer *jitterbuffer;
GstRtpJitterBufferPrivate *priv;
g_return_if_fail (GST_IS_RTP_JITTER_BUFFER (element));
g_return_if_fail (GST_IS_PAD (pad));
jitterbuffer = GST_RTP_JITTER_BUFFER (element);
priv = jitterbuffer->priv;
GST_DEBUG_OBJECT (element, "releasing pad %s:%s", GST_DEBUG_PAD_NAME (pad));
if (priv->rtcpsinkpad == pad) {
remove_rtcp_sink (jitterbuffer);
} else
goto wrong_pad;
return;
/* ERRORS */
wrong_pad:
{
g_warning ("gstjitterbuffer: asked to release an unknown pad");
return;
}
}
static void
gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer)
{
GstRtpJitterBufferPrivate *priv;
priv = jitterbuffer->priv;
/* this will trigger a new pt-map request signal, FIXME, do something better. */
JBUF_LOCK (priv);
priv->clock_rate = -1;
JBUF_UNLOCK (priv);
}
static GstClockTime
gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jbuf, gboolean active,
guint64 offset)
{
GstRtpJitterBufferPrivate *priv;
GstClockTime last_out;
GstBuffer *head;
priv = jbuf->priv;
JBUF_LOCK (priv);
GST_DEBUG_OBJECT (jbuf, "setting active %d with offset %" GST_TIME_FORMAT,
active, GST_TIME_ARGS (offset));
if (active != priv->active) {
/* add the amount of time spent in paused to the output offset. All
* outgoing buffers will have this offset applied to their timestamps in
* order to make them arrive in time in the sink. */
priv->out_offset = offset;
GST_DEBUG_OBJECT (jbuf, "out offset %" GST_TIME_FORMAT,
GST_TIME_ARGS (priv->out_offset));
priv->active = active;
JBUF_SIGNAL (priv);
}
if (!active) {
rtp_jitter_buffer_set_buffering (priv->jbuf, TRUE);
}
if ((head = rtp_jitter_buffer_peek (priv->jbuf))) {
/* head buffer timestamp and offset gives our output time */
last_out = GST_BUFFER_TIMESTAMP (head) + priv->ts_offset;
} else {
/* use last known time when the buffer is empty */
last_out = priv->last_out_time;
}
JBUF_UNLOCK (priv);
return last_out;
}
static GstCaps *
gst_rtp_jitter_buffer_getcaps (GstPad * pad)
{
GstRtpJitterBuffer *jitterbuffer;
GstRtpJitterBufferPrivate *priv;
GstPad *other;
GstCaps *caps;
const GstCaps *templ;
jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
priv = jitterbuffer->priv;
other = (pad == priv->srcpad ? priv->sinkpad : priv->srcpad);
caps = gst_pad_peer_get_caps (other);
templ = gst_pad_get_pad_template_caps (pad);
if (caps == NULL) {
GST_DEBUG_OBJECT (jitterbuffer, "copy template");
caps = gst_caps_copy (templ);
} else {
GstCaps *intersect;
GST_DEBUG_OBJECT (jitterbuffer, "intersect with template");
intersect = gst_caps_intersect (caps, templ);
gst_caps_unref (caps);
caps = intersect;
}
gst_object_unref (jitterbuffer);
return caps;
}
/*
* Must be called with JBUF_LOCK held
*/
static gboolean
gst_jitter_buffer_sink_parse_caps (GstRtpJitterBuffer * jitterbuffer,
GstCaps * caps)
{
GstRtpJitterBufferPrivate *priv;
GstStructure *caps_struct;
guint val;
GstClockTime tval;
priv = jitterbuffer->priv;
/* first parse the caps */
caps_struct = gst_caps_get_structure (caps, 0);
GST_DEBUG_OBJECT (jitterbuffer, "got caps");
/* we need a clock-rate to convert the rtp timestamps to GStreamer time and to
* measure the amount of data in the buffer */
if (!gst_structure_get_int (caps_struct, "clock-rate", &priv->clock_rate))
goto error;
if (priv->clock_rate <= 0)
goto wrong_rate;
GST_DEBUG_OBJECT (jitterbuffer, "got clock-rate %d", priv->clock_rate);
/* The clock base is the RTP timestamp corrsponding to the npt-start value. We
* can use this to track the amount of time elapsed on the sender. */
if (gst_structure_get_uint (caps_struct, "clock-base", &val))
priv->clock_base = val;
else
priv->clock_base = -1;
priv->ext_timestamp = priv->clock_base;
GST_DEBUG_OBJECT (jitterbuffer, "got clock-base %" G_GINT64_FORMAT,
priv->clock_base);
if (gst_structure_get_uint (caps_struct, "seqnum-base", &val)) {
/* first expected seqnum, only update when we didn't have a previous base. */
if (priv->next_in_seqnum == -1)
priv->next_in_seqnum = val;
if (priv->next_seqnum == -1)
priv->next_seqnum = val;
}
GST_DEBUG_OBJECT (jitterbuffer, "got seqnum-base %d", priv->next_in_seqnum);
/* the start and stop times. The seqnum-base corresponds to the start time. We
* will keep track of the seqnums on the output and when we reach the one
* corresponding to npt-stop, we emit the npt-stop-reached signal */
if (gst_structure_get_clock_time (caps_struct, "npt-start", &tval))
priv->npt_start = tval;
else
priv->npt_start = 0;
if (gst_structure_get_clock_time (caps_struct, "npt-stop", &tval))
priv->npt_stop = tval;
else
priv->npt_stop = -1;
GST_DEBUG_OBJECT (jitterbuffer,
"npt start/stop: %" GST_TIME_FORMAT "-%" GST_TIME_FORMAT,
GST_TIME_ARGS (priv->npt_start), GST_TIME_ARGS (priv->npt_stop));
return TRUE;
/* ERRORS */
error:
{
GST_DEBUG_OBJECT (jitterbuffer, "No clock-rate in caps!");
return FALSE;
}
wrong_rate:
{
GST_DEBUG_OBJECT (jitterbuffer, "Invalid clock-rate %d", priv->clock_rate);
return FALSE;
}
}
static gboolean
gst_jitter_buffer_sink_setcaps (GstPad * pad, GstCaps * caps)
{
GstRtpJitterBuffer *jitterbuffer;
GstRtpJitterBufferPrivate *priv;
gboolean res;
jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
priv = jitterbuffer->priv;
JBUF_LOCK (priv);
res = gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
JBUF_UNLOCK (priv);
/* set same caps on srcpad on success */
if (res)
gst_pad_set_caps (priv->srcpad, caps);
gst_object_unref (jitterbuffer);
return res;
}
static void
gst_rtp_jitter_buffer_flush_start (GstRtpJitterBuffer * jitterbuffer)
{
GstRtpJitterBufferPrivate *priv;
priv = jitterbuffer->priv;
JBUF_LOCK (priv);
/* mark ourselves as flushing */
priv->srcresult = GST_FLOW_WRONG_STATE;
GST_DEBUG_OBJECT (jitterbuffer, "Disabling pop on queue");
/* this unblocks any waiting pops on the src pad task */
JBUF_SIGNAL (priv);
/* unlock clock, we just unschedule, the entry will be released by the
* locking streaming thread. */
if (priv->clock_id) {
gst_clock_id_unschedule (priv->clock_id);
priv->unscheduled = TRUE;
}
JBUF_UNLOCK (priv);
}
static void
gst_rtp_jitter_buffer_flush_stop (GstRtpJitterBuffer * jitterbuffer)
{
GstRtpJitterBufferPrivate *priv;
priv = jitterbuffer->priv;
JBUF_LOCK (priv);
GST_DEBUG_OBJECT (jitterbuffer, "Enabling pop on queue");
/* Mark as non flushing */
priv->srcresult = GST_FLOW_OK;
gst_segment_init (&priv->segment, GST_FORMAT_TIME);
priv->last_popped_seqnum = -1;
priv->last_out_time = -1;
priv->next_seqnum = -1;
priv->next_in_seqnum = -1;
priv->clock_rate = -1;
priv->eos = FALSE;
priv->estimated_eos = -1;
priv->last_elapsed = 0;
priv->reached_npt_stop = FALSE;
priv->ext_timestamp = -1;
GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
rtp_jitter_buffer_flush (priv->jbuf);
rtp_jitter_buffer_reset_skew (priv->jbuf);
JBUF_UNLOCK (priv);
}
static gboolean
gst_rtp_jitter_buffer_src_activate_push (GstPad * pad, gboolean active)
{
gboolean result = TRUE;
GstRtpJitterBuffer *jitterbuffer = NULL;
jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
if (active) {
/* allow data processing */
gst_rtp_jitter_buffer_flush_stop (jitterbuffer);
/* start pushing out buffers */
GST_DEBUG_OBJECT (jitterbuffer, "Starting task on srcpad");
gst_pad_start_task (jitterbuffer->priv->srcpad,
(GstTaskFunction) gst_rtp_jitter_buffer_loop, jitterbuffer);
} else {
/* make sure all data processing stops ASAP */
gst_rtp_jitter_buffer_flush_start (jitterbuffer);
/* NOTE this will hardlock if the state change is called from the src pad
* task thread because we will _join() the thread. */
GST_DEBUG_OBJECT (jitterbuffer, "Stopping task on srcpad");
result = gst_pad_stop_task (pad);
}
gst_object_unref (jitterbuffer);
return result;
}
static GstStateChangeReturn
gst_rtp_jitter_buffer_change_state (GstElement * element,
GstStateChange transition)
{
GstRtpJitterBuffer *jitterbuffer;
GstRtpJitterBufferPrivate *priv;
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
jitterbuffer = GST_RTP_JITTER_BUFFER (element);
priv = jitterbuffer->priv;
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
JBUF_LOCK (priv);
/* reset negotiated values */
priv->clock_rate = -1;
priv->clock_base = -1;
priv->peer_latency = 0;
priv->last_pt = -1;
/* block until we go to PLAYING */
priv->blocked = TRUE;
JBUF_UNLOCK (priv);
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
JBUF_LOCK (priv);
/* unblock to allow streaming in PLAYING */
priv->blocked = FALSE;
JBUF_SIGNAL (priv);
JBUF_UNLOCK (priv);
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
/* we are a live element because we sync to the clock, which we can only
* do in the PLAYING state */
if (ret != GST_STATE_CHANGE_FAILURE)
ret = GST_STATE_CHANGE_NO_PREROLL;
break;
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
JBUF_LOCK (priv);
/* block to stop streaming when PAUSED */
priv->blocked = TRUE;
JBUF_UNLOCK (priv);
if (ret != GST_STATE_CHANGE_FAILURE)
ret = GST_STATE_CHANGE_NO_PREROLL;
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_NULL:
break;
default:
break;
}
return ret;
}
static gboolean
gst_rtp_jitter_buffer_src_event (GstPad * pad, GstEvent * event)
{
gboolean ret = TRUE;
GstRtpJitterBuffer *jitterbuffer;
GstRtpJitterBufferPrivate *priv;
jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
priv = jitterbuffer->priv;
GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_LATENCY:
{
GstClockTime latency;
gst_event_parse_latency (event, &latency);
JBUF_LOCK (priv);
/* adjust the overall buffer delay to the total pipeline latency in
* buffering mode because if downstream consumes too fast (because of
* large latency or queues, we would start rebuffering again. */
if (rtp_jitter_buffer_get_mode (priv->jbuf) ==
RTP_JITTER_BUFFER_MODE_BUFFER) {
rtp_jitter_buffer_set_delay (priv->jbuf, latency);
}
JBUF_UNLOCK (priv);
ret = gst_pad_push_event (priv->sinkpad, event);
break;
}
default:
ret = gst_pad_push_event (priv->sinkpad, event);
break;
}
gst_object_unref (jitterbuffer);
return ret;
}
static gboolean
gst_rtp_jitter_buffer_sink_event (GstPad * pad, GstEvent * event)
{
gboolean ret = TRUE;
GstRtpJitterBuffer *jitterbuffer;
GstRtpJitterBufferPrivate *priv;
jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
priv = jitterbuffer->priv;
GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_NEWSEGMENT:
{
GstFormat format;
gdouble rate, arate;
gint64 start, stop, time;
gboolean update;
gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
&start, &stop, &time);
/* we need time for now */
if (format != GST_FORMAT_TIME)
goto newseg_wrong_format;
GST_DEBUG_OBJECT (jitterbuffer,
"newsegment: update %d, rate %g, arate %g, start %" GST_TIME_FORMAT
", stop %" GST_TIME_FORMAT ", time %" GST_TIME_FORMAT,
update, rate, arate, GST_TIME_ARGS (start), GST_TIME_ARGS (stop),
GST_TIME_ARGS (time));
/* now configure the values, we need these to time the release of the
* buffers on the srcpad. */
gst_segment_set_newsegment_full (&priv->segment, update,
rate, arate, format, start, stop, time);
/* FIXME, push SEGMENT in the queue. Sorting order might be difficult. */
ret = gst_pad_push_event (priv->srcpad, event);
break;
}
case GST_EVENT_FLUSH_START:
gst_rtp_jitter_buffer_flush_start (jitterbuffer);
ret = gst_pad_push_event (priv->srcpad, event);
break;
case GST_EVENT_FLUSH_STOP:
ret = gst_pad_push_event (priv->srcpad, event);
ret = gst_rtp_jitter_buffer_src_activate_push (priv->srcpad, TRUE);
break;
case GST_EVENT_EOS:
{
/* push EOS in queue. We always push it at the head */
JBUF_LOCK (priv);
/* check for flushing, we need to discard the event and return FALSE when
* we are flushing */
ret = priv->srcresult == GST_FLOW_OK;
if (ret && !priv->eos) {
GST_DEBUG_OBJECT (jitterbuffer, "queuing EOS");
priv->eos = TRUE;
JBUF_SIGNAL (priv);
} else if (priv->eos) {
GST_DEBUG_OBJECT (jitterbuffer, "dropping EOS, we are already EOS");
} else {
GST_DEBUG_OBJECT (jitterbuffer, "dropping EOS, reason %s",
gst_flow_get_name (priv->srcresult));
}
JBUF_UNLOCK (priv);
gst_event_unref (event);
break;
}
default:
ret = gst_pad_push_event (priv->srcpad, event);
break;
}
done:
gst_object_unref (jitterbuffer);
return ret;
/* ERRORS */
newseg_wrong_format:
{
GST_DEBUG_OBJECT (jitterbuffer, "received non TIME newsegment");
ret = FALSE;
goto done;
}
}
static gboolean
gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad, GstEvent * event)
{
GstRtpJitterBuffer *jitterbuffer;
GstRtpJitterBufferPrivate *priv;
jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
priv = jitterbuffer->priv;
GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_START:
break;
case GST_EVENT_FLUSH_STOP:
break;
default:
break;
}
gst_event_unref (event);
gst_object_unref (jitterbuffer);
return TRUE;
}
/*
* Must be called with JBUF_LOCK held, will release the LOCK when emiting the
* signal. The function returns GST_FLOW_ERROR when a parsing error happened and
* GST_FLOW_WRONG_STATE when the element is shutting down. On success
* GST_FLOW_OK is returned.
*/
static GstFlowReturn
gst_rtp_jitter_buffer_get_clock_rate (GstRtpJitterBuffer * jitterbuffer,
guint8 pt)
{
GValue ret = { 0 };
GValue args[2] = { {0}, {0} };
GstCaps *caps;
gboolean res;
g_value_init (&args[0], GST_TYPE_ELEMENT);
g_value_set_object (&args[0], jitterbuffer);
g_value_init (&args[1], G_TYPE_UINT);
g_value_set_uint (&args[1], pt);
g_value_init (&ret, GST_TYPE_CAPS);
g_value_set_boxed (&ret, NULL);
JBUF_UNLOCK (jitterbuffer->priv);
g_signal_emitv (args, gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP], 0,
&ret);
JBUF_LOCK_CHECK (jitterbuffer->priv, out_flushing);
g_value_unset (&args[0]);
g_value_unset (&args[1]);
caps = (GstCaps *) g_value_dup_boxed (&ret);
g_value_unset (&ret);
if (!caps)
goto no_caps;
res = gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
gst_caps_unref (caps);
if (G_UNLIKELY (!res))
goto parse_failed;
return GST_FLOW_OK;
/* ERRORS */
no_caps:
{
GST_DEBUG_OBJECT (jitterbuffer, "could not get caps");
return GST_FLOW_ERROR;
}
out_flushing:
{
GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
return GST_FLOW_WRONG_STATE;
}
parse_failed:
{
GST_DEBUG_OBJECT (jitterbuffer, "parse failed");
return GST_FLOW_ERROR;
}
}
static void
post_buffering_percent (GstRtpJitterBuffer * jitterbuffer, gint percent)
{
GstMessage *message;
/* Post a buffering message */
message = gst_message_new_buffering (GST_OBJECT_CAST (jitterbuffer), percent);
gst_message_set_buffering_stats (message, GST_BUFFERING_LIVE, -1, -1, -1);
gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), message);
}
static GstFlowReturn
gst_rtp_jitter_buffer_chain (GstPad * pad, GstBuffer * buffer)
{
GstRtpJitterBuffer *jitterbuffer;
GstRtpJitterBufferPrivate *priv;
guint16 seqnum;
GstFlowReturn ret = GST_FLOW_OK;
GstClockTime timestamp;
guint64 latency_ts;
gboolean tail;
gint percent = -1;
guint8 pt;
jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
if (G_UNLIKELY (!gst_rtp_buffer_validate (buffer)))
goto invalid_buffer;
priv = jitterbuffer->priv;
pt = gst_rtp_buffer_get_payload_type (buffer);
/* take the timestamp of the buffer. This is the time when the packet was
* received and is used to calculate jitter and clock skew. We will adjust
* this timestamp with the smoothed value after processing it in the
* jitterbuffer. */
timestamp = GST_BUFFER_TIMESTAMP (buffer);
/* bring to running time */
timestamp = gst_segment_to_running_time (&priv->segment, GST_FORMAT_TIME,
timestamp);
seqnum = gst_rtp_buffer_get_seq (buffer);
GST_DEBUG_OBJECT (jitterbuffer,
"Received packet #%d at time %" GST_TIME_FORMAT, seqnum,
GST_TIME_ARGS (timestamp));
JBUF_LOCK_CHECK (priv, out_flushing);
if (G_UNLIKELY (priv->last_pt != pt)) {
GstCaps *caps;
GST_DEBUG_OBJECT (jitterbuffer, "pt changed from %u to %u", priv->last_pt,
pt);
priv->last_pt = pt;
/* reset clock-rate so that we get a new one */
priv->clock_rate = -1;
/* Try to get the clock-rate from the caps first if we can. If there are no
* caps we must fire the signal to get the clock-rate. */
if ((caps = GST_BUFFER_CAPS (buffer))) {
gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
}
}
if (G_UNLIKELY (priv->clock_rate == -1)) {
/* no clock rate given on the caps, try to get one with the signal */
if (gst_rtp_jitter_buffer_get_clock_rate (jitterbuffer,
pt) == GST_FLOW_WRONG_STATE)
goto out_flushing;
if (G_UNLIKELY (priv->clock_rate == -1))
goto no_clock_rate;
}
/* don't accept more data on EOS */
if (G_UNLIKELY (priv->eos))
goto have_eos;
/* now check against our expected seqnum */
if (G_LIKELY (priv->next_in_seqnum != -1)) {
gint gap;
gboolean reset = FALSE;
gap = gst_rtp_buffer_compare_seqnum (priv->next_in_seqnum, seqnum);
if (G_UNLIKELY (gap != 0)) {
GST_DEBUG_OBJECT (jitterbuffer, "expected #%d, got #%d, gap of %d",
priv->next_in_seqnum, seqnum, gap);
/* priv->next_in_seqnum >= seqnum, this packet is too late or the
* sender might have been restarted with different seqnum. */
if (gap < -RTP_MAX_MISORDER) {
GST_DEBUG_OBJECT (jitterbuffer, "reset: buffer too old %d", gap);
reset = TRUE;
}
/* priv->next_in_seqnum < seqnum, this is a new packet */
else if (G_UNLIKELY (gap > RTP_MAX_DROPOUT)) {
GST_DEBUG_OBJECT (jitterbuffer, "reset: too many dropped packets %d",
gap);
reset = TRUE;
} else {
GST_DEBUG_OBJECT (jitterbuffer, "tolerable gap");
}
}
if (G_UNLIKELY (reset)) {
GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
rtp_jitter_buffer_flush (priv->jbuf);
rtp_jitter_buffer_reset_skew (priv->jbuf);
priv->last_popped_seqnum = -1;
priv->next_seqnum = seqnum;
}
}
priv->next_in_seqnum = (seqnum + 1) & 0xffff;
/* let's check if this buffer is too late, we can only accept packets with
* bigger seqnum than the one we last pushed. */
if (G_LIKELY (priv->last_popped_seqnum != -1)) {
gint gap;
gap = gst_rtp_buffer_compare_seqnum (priv->last_popped_seqnum, seqnum);
/* priv->last_popped_seqnum >= seqnum, we're too late. */
if (G_UNLIKELY (gap <= 0))
goto too_late;
}
/* let's drop oldest packet if the queue is already full and drop-on-latency
* is set. We can only do this when there actually is a latency. When no
* latency is set, we just pump it in the queue and let the other end push it
* out as fast as possible. */
if (priv->latency_ms && priv->drop_on_latency) {
latency_ts =
gst_util_uint64_scale_int (priv->latency_ms, priv->clock_rate, 1000);
if (G_UNLIKELY (rtp_jitter_buffer_get_ts_diff (priv->jbuf) >= latency_ts)) {
GstBuffer *old_buf;
old_buf = rtp_jitter_buffer_pop (priv->jbuf, &percent);
GST_DEBUG_OBJECT (jitterbuffer, "Queue full, dropping old packet #%d",
gst_rtp_buffer_get_seq (old_buf));
gst_buffer_unref (old_buf);
}
}
/* we need to make the metadata writable before pushing it in the jitterbuffer
* because the jitterbuffer will update the timestamp */
buffer = gst_buffer_make_metadata_writable (buffer);
/* now insert the packet into the queue in sorted order. This function returns
* FALSE if a packet with the same seqnum was already in the queue, meaning we
* have a duplicate. */
if (G_UNLIKELY (!rtp_jitter_buffer_insert (priv->jbuf, buffer, timestamp,
priv->clock_rate, &tail, &percent)))
goto duplicate;
/* signal addition of new buffer when the _loop is waiting. */
if (priv->waiting)
JBUF_SIGNAL (priv);
/* let's unschedule and unblock any waiting buffers. We only want to do this
* when the tail buffer changed */
if (G_UNLIKELY (priv->clock_id && tail)) {
GST_DEBUG_OBJECT (jitterbuffer,
"Unscheduling waiting buffer, new tail buffer");
gst_clock_id_unschedule (priv->clock_id);
priv->unscheduled = TRUE;
}
GST_DEBUG_OBJECT (jitterbuffer, "Pushed packet #%d, now %d packets, tail: %d",
seqnum, rtp_jitter_buffer_num_packets (priv->jbuf), tail);
finished:
JBUF_UNLOCK (priv);
if (percent != -1)
post_buffering_percent (jitterbuffer, percent);
gst_object_unref (jitterbuffer);
return ret;
/* ERRORS */
invalid_buffer:
{
/* this is not fatal but should be filtered earlier */
GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
("Received invalid RTP payload, dropping"));
gst_buffer_unref (buffer);
gst_object_unref (jitterbuffer);
return GST_FLOW_OK;
}
no_clock_rate:
{
GST_WARNING_OBJECT (jitterbuffer,
"No clock-rate in caps!, dropping buffer");
gst_buffer_unref (buffer);
goto finished;
}
out_flushing:
{
ret = priv->srcresult;
GST_DEBUG_OBJECT (jitterbuffer, "flushing %s", gst_flow_get_name (ret));
gst_buffer_unref (buffer);
goto finished;
}
have_eos:
{
ret = GST_FLOW_UNEXPECTED;
GST_WARNING_OBJECT (jitterbuffer, "we are EOS, refusing buffer");
gst_buffer_unref (buffer);
goto finished;
}
too_late:
{
GST_WARNING_OBJECT (jitterbuffer, "Packet #%d too late as #%d was already"
" popped, dropping", seqnum, priv->last_popped_seqnum);
priv->num_late++;
gst_buffer_unref (buffer);
goto finished;
}
duplicate:
{
GST_WARNING_OBJECT (jitterbuffer, "Duplicate packet #%d detected, dropping",
seqnum);
priv->num_duplicates++;
gst_buffer_unref (buffer);
goto finished;
}
}
static GstClockTime
apply_offset (GstRtpJitterBuffer * jitterbuffer, GstClockTime timestamp)
{
GstRtpJitterBufferPrivate *priv;
priv = jitterbuffer->priv;
if (timestamp == -1)
return -1;
/* apply the timestamp offset, this is used for inter stream sync */
timestamp += priv->ts_offset;
/* add the offset, this is used when buffering */
timestamp += priv->out_offset;
return timestamp;
}
static GstClockTime
get_sync_time (GstRtpJitterBuffer * jitterbuffer, GstClockTime timestamp)
{
GstClockTime result;
GstRtpJitterBufferPrivate *priv;
priv = jitterbuffer->priv;
result = timestamp + GST_ELEMENT_CAST (jitterbuffer)->base_time;
/* add latency, this includes our own latency and the peer latency. */
result += priv->latency_ns;
result += priv->peer_latency;
return result;
}
static gboolean
eos_reached (GstClock * clock, GstClockTime time, GstClockID id,
GstRtpJitterBuffer * jitterbuffer)
{
GstRtpJitterBufferPrivate *priv;
priv = jitterbuffer->priv;
JBUF_LOCK_CHECK (priv, flushing);
if (priv->waiting) {
GST_DEBUG_OBJECT (jitterbuffer, "got the NPT timeout");
priv->reached_npt_stop = TRUE;
JBUF_SIGNAL (priv);
}
JBUF_UNLOCK (priv);
return TRUE;
/* ERRORS */
flushing:
{
JBUF_UNLOCK (priv);
return FALSE;
}
}
/*
* This funcion will push out buffers on the source pad.
*
* For each pushed buffer, the seqnum is recorded, if the next buffer B has a
* different seqnum (missing packets before B), this function will wait for the
* missing packet to arrive up to the timestamp of buffer B.
*/
static void
gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer)
{
GstRtpJitterBufferPrivate *priv;
GstBuffer *outbuf;
GstFlowReturn result;
guint16 seqnum;
guint32 next_seqnum;
GstClockTime timestamp, out_time;
gboolean discont = FALSE;
gint gap;
GstClock *clock;
GstClockID id;
GstClockTime sync_time;
gint percent = -1;
priv = jitterbuffer->priv;
JBUF_LOCK_CHECK (priv, flushing);
again:
GST_DEBUG_OBJECT (jitterbuffer, "Peeking item");
while (TRUE) {
id = NULL;
/* always wait if we are blocked */
if (G_LIKELY (!priv->blocked)) {
/* we're buffering but not EOS, wait. */
if (!priv->eos && (!priv->active
|| rtp_jitter_buffer_is_buffering (priv->jbuf)))
goto do_wait;
/* if we have a packet, we can exit the loop and grab it */
if (rtp_jitter_buffer_num_packets (priv->jbuf) > 0)
break;
/* no packets but we are EOS, do eos logic */
if (G_UNLIKELY (priv->eos))
goto do_eos;
/* underrun, wait for packets or flushing now if we are expecting an EOS
* timeout, set the async timer for it too */
if (priv->estimated_eos != -1 && !priv->reached_npt_stop) {
sync_time = get_sync_time (jitterbuffer, priv->estimated_eos);
GST_OBJECT_LOCK (jitterbuffer);
clock = GST_ELEMENT_CLOCK (jitterbuffer);
if (clock) {
GST_DEBUG_OBJECT (jitterbuffer, "scheduling timeout");
id = gst_clock_new_single_shot_id (clock, sync_time);
gst_clock_id_wait_async (id, (GstClockCallback) eos_reached,
jitterbuffer);
}
GST_OBJECT_UNLOCK (jitterbuffer);
}
}
do_wait:
/* now we wait */
GST_DEBUG_OBJECT (jitterbuffer, "waiting");
priv->waiting = TRUE;
JBUF_WAIT (priv);
priv->waiting = FALSE;
GST_DEBUG_OBJECT (jitterbuffer, "waiting done");
if (id) {
/* unschedule any pending async notifications we might have */
gst_clock_id_unschedule (id);
gst_clock_id_unref (id);
}
if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK))
goto flushing;
if (id && priv->reached_npt_stop) {
goto do_npt_stop;
}
}
/* peek a buffer, we're just looking at the timestamp and the sequence number.
* If all is fine, we'll pop and push it. If the sequence number is wrong we
* wait on the timestamp. In the chain function we will unlock the wait when a
* new buffer is available. The peeked buffer is valid for as long as we hold
* the jitterbuffer lock. */
outbuf = rtp_jitter_buffer_peek (priv->jbuf);
/* get the seqnum and the next expected seqnum */
seqnum = gst_rtp_buffer_get_seq (outbuf);
next_seqnum = priv->next_seqnum;
/* get the timestamp, this is already corrected for clock skew by the
* jitterbuffer */
timestamp = GST_BUFFER_TIMESTAMP (outbuf);
GST_DEBUG_OBJECT (jitterbuffer,
"Peeked buffer #%d, expect #%d, timestamp %" GST_TIME_FORMAT
", now %d left", seqnum, next_seqnum, GST_TIME_ARGS (timestamp),
rtp_jitter_buffer_num_packets (priv->jbuf));
/* apply our timestamp offset to the incomming buffer, this will be our output
* timestamp. */
out_time = apply_offset (jitterbuffer, timestamp);
/* get the gap between this and the previous packet. If we don't know the
* previous packet seqnum assume no gap. */
if (G_LIKELY (next_seqnum != -1)) {
gap = gst_rtp_buffer_compare_seqnum (next_seqnum, seqnum);
/* if we have a packet that we already pushed or considered dropped, pop it
* off and get the next packet */
if (G_UNLIKELY (gap < 0)) {
GST_DEBUG_OBJECT (jitterbuffer, "Old packet #%d, next #%d dropping",
seqnum, next_seqnum);
outbuf = rtp_jitter_buffer_pop (priv->jbuf, &percent);
gst_buffer_unref (outbuf);
goto again;
}
} else {
GST_DEBUG_OBJECT (jitterbuffer, "no next seqnum known, first packet");
gap = -1;
}
/* If we don't know what the next seqnum should be (== -1) we have to wait
* because it might be possible that we are not receiving this buffer in-order,
* a buffer with a lower seqnum could arrive later and we want to push that
* earlier buffer before this buffer then.
* If we know the expected seqnum, we can compare it to the current seqnum to
* determine if we have missing a packet. If we have a missing packet (which
* must be before this packet) we can wait for it until the deadline for this
* packet expires. */
if (G_UNLIKELY (gap != 0 && out_time != -1)) {
GstClockReturn ret;
GstClockTime duration = GST_CLOCK_TIME_NONE;
if (gap > 0) {
/* we have a gap */
GST_DEBUG_OBJECT (jitterbuffer,
"Sequence number GAP detected: expected %d instead of %d (%d missing)",
next_seqnum, seqnum, gap);
if (priv->last_out_time != -1) {
GST_DEBUG_OBJECT (jitterbuffer,
"out_time %" GST_TIME_FORMAT ", last %" GST_TIME_FORMAT,
GST_TIME_ARGS (out_time), GST_TIME_ARGS (priv->last_out_time));
/* interpolate between the current time and the last time based on
* number of packets we are missing, this is the estimated duration
* for the missing packet based on equidistant packet spacing. Also make
* sure we never go negative. */
if (out_time >= priv->last_out_time)
duration = (out_time - priv->last_out_time) / (gap + 1);
else
goto lost;
GST_DEBUG_OBJECT (jitterbuffer, "duration %" GST_TIME_FORMAT,
GST_TIME_ARGS (duration));
/* add this duration to the timestamp of the last packet we pushed */
out_time = (priv->last_out_time + duration);
}
} else {
/* we don't know what the next_seqnum should be, wait for the last
* possible moment to push this buffer, maybe we get an earlier seqnum
* while we wait */
GST_DEBUG_OBJECT (jitterbuffer, "First buffer %d, do sync", seqnum);
}
GST_OBJECT_LOCK (jitterbuffer);
clock = GST_ELEMENT_CLOCK (jitterbuffer);
if (!clock) {
GST_OBJECT_UNLOCK (jitterbuffer);
/* let's just push if there is no clock */
GST_DEBUG_OBJECT (jitterbuffer, "No clock, push right away");
goto push_buffer;
}
GST_DEBUG_OBJECT (jitterbuffer, "sync to timestamp %" GST_TIME_FORMAT,
GST_TIME_ARGS (out_time));
/* prepare for sync against clock */
sync_time = get_sync_time (jitterbuffer, out_time);
/* create an entry for the clock */
id = priv->clock_id = gst_clock_new_single_shot_id (clock, sync_time);
priv->unscheduled = FALSE;
GST_OBJECT_UNLOCK (jitterbuffer);
/* release the lock so that the other end can push stuff or unlock */
JBUF_UNLOCK (priv);
ret = gst_clock_id_wait (id, NULL);
JBUF_LOCK (priv);
/* and free the entry */
gst_clock_id_unref (id);
priv->clock_id = NULL;
/* at this point, the clock could have been unlocked by a timeout, a new
* tail element was added to the queue or because we are shutting down. Check
* for shutdown first. */
if G_UNLIKELY
((priv->srcresult != GST_FLOW_OK))
goto flushing;
/* if we got unscheduled and we are not flushing, it's because a new tail
* element became available in the queue or we flushed the queue.
* Grab it and try to push or sync. */
if (ret == GST_CLOCK_UNSCHEDULED || priv->unscheduled) {
GST_DEBUG_OBJECT (jitterbuffer,
"Wait got unscheduled, will retry to push with new buffer");
goto again;
}
lost:
/* we now timed out, this means we lost a packet or finished synchronizing
* on the first buffer. */
if (gap > 0) {
GstEvent *event;
/* we had a gap and thus we lost a packet. Create an event for this. */
GST_WARNING_OBJECT (jitterbuffer, "Packet #%d lost", next_seqnum);
priv->num_late++;
discont = TRUE;
/* update our expected next packet */
priv->last_popped_seqnum = next_seqnum;
priv->last_out_time = out_time;
priv->next_seqnum = (next_seqnum + 1) & 0xffff;
if (priv->do_lost) {
/* create paket lost event */
event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM,
gst_structure_new ("GstRTPPacketLost",
"seqnum", G_TYPE_UINT, (guint) next_seqnum,
"timestamp", G_TYPE_UINT64, out_time,
"duration", G_TYPE_UINT64, duration, NULL));
JBUF_UNLOCK (priv);
gst_pad_push_event (priv->srcpad, event);
JBUF_LOCK_CHECK (priv, flushing);
}
/* look for next packet */
goto again;
}
/* there was no known gap,just the first packet, exit the loop and push */
GST_DEBUG_OBJECT (jitterbuffer, "First packet #%d synced", seqnum);
/* get new timestamp, latency might have changed */
out_time = apply_offset (jitterbuffer, timestamp);
}
push_buffer:
/* when we get here we are ready to pop and push the buffer */
outbuf = rtp_jitter_buffer_pop (priv->jbuf, &percent);
if (G_UNLIKELY (discont || priv->discont)) {
/* set DISCONT flag when we missed a packet. We pushed the buffer writable
* into the jitterbuffer so we can modify now. */
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
priv->discont = FALSE;
}
/* apply timestamp with offset to buffer now */
GST_BUFFER_TIMESTAMP (outbuf) = out_time;
/* update the elapsed time when we need to check against the npt stop time. */
if (priv->npt_stop != -1 && priv->ext_timestamp != -1
&& priv->clock_base != -1 && priv->clock_rate > 0) {
guint64 ext_time, elapsed, estimated;
guint32 rtp_time;
rtp_time = gst_rtp_buffer_get_timestamp (outbuf);
GST_LOG_OBJECT (jitterbuffer, "rtp %" G_GUINT32_FORMAT ", ext %"
G_GUINT64_FORMAT, rtp_time, priv->ext_timestamp);
if (rtp_time < priv->ext_timestamp) {
ext_time = priv->ext_timestamp;
} else {
ext_time = gst_rtp_buffer_ext_timestamp (&priv->ext_timestamp, rtp_time);
}
if (ext_time > priv->clock_base)
elapsed = ext_time - priv->clock_base;
else
elapsed = 0;
elapsed = gst_util_uint64_scale_int (elapsed, GST_SECOND, priv->clock_rate);
if (elapsed > priv->last_elapsed) {
guint64 left;
priv->last_elapsed = elapsed;
left = priv->npt_stop - priv->npt_start;
if (elapsed > 0)
estimated = gst_util_uint64_scale (out_time, left, elapsed);
else
estimated = -1;
GST_LOG_OBJECT (jitterbuffer, "elapsed %" GST_TIME_FORMAT ", estimated %"
GST_TIME_FORMAT, GST_TIME_ARGS (elapsed), GST_TIME_ARGS (estimated));
priv->estimated_eos = estimated;
}
}
/* now we are ready to push the buffer. Save the seqnum and release the lock
* so the other end can push stuff in the queue again. */
priv->last_popped_seqnum = seqnum;
priv->last_out_time = out_time;
priv->next_seqnum = (seqnum + 1) & 0xffff;
JBUF_UNLOCK (priv);
if (percent != -1)
post_buffering_percent (jitterbuffer, percent);
/* push buffer */
GST_DEBUG_OBJECT (jitterbuffer,
"Pushing buffer %d, timestamp %" GST_TIME_FORMAT, seqnum,
GST_TIME_ARGS (out_time));
result = gst_pad_push (priv->srcpad, outbuf);
if (G_UNLIKELY (result != GST_FLOW_OK))
goto pause;
return;
/* ERRORS */
do_eos:
{
/* store result, we are flushing now */
GST_DEBUG_OBJECT (jitterbuffer, "We are EOS, pushing EOS downstream");
priv->srcresult = GST_FLOW_UNEXPECTED;
gst_pad_pause_task (priv->srcpad);
JBUF_UNLOCK (priv);
gst_pad_push_event (priv->srcpad, gst_event_new_eos ());
return;
}
do_npt_stop:
{
/* store result, we are flushing now */
GST_DEBUG_OBJECT (jitterbuffer, "We reached the NPT stop");
JBUF_UNLOCK (priv);
g_signal_emit (jitterbuffer,
gst_rtp_jitter_buffer_signals[SIGNAL_ON_NPT_STOP], 0, NULL);
return;
}
flushing:
{
GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
gst_pad_pause_task (priv->srcpad);
JBUF_UNLOCK (priv);
return;
}
pause:
{
GST_DEBUG_OBJECT (jitterbuffer, "pausing task, reason %s",
gst_flow_get_name (result));
JBUF_LOCK (priv);
/* store result */
priv->srcresult = result;
/* we don't post errors or anything because upstream will do that for us
* when we pass the return value upstream. */
gst_pad_pause_task (priv->srcpad);
JBUF_UNLOCK (priv);
return;
}
}
static GstFlowReturn
gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad, GstBuffer * buffer)
{
GstRtpJitterBuffer *jitterbuffer;
GstRtpJitterBufferPrivate *priv;
GstFlowReturn ret = GST_FLOW_OK;
guint64 base_rtptime, timestamp;
guint32 clock_rate;
guint64 last_rtptime;
guint32 ssrc;
GstRTCPPacket packet;
guint64 ext_rtptime, diff;
guint32 rtptime;
gboolean drop = FALSE;
jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
if (G_UNLIKELY (!gst_rtcp_buffer_validate (buffer)))
goto invalid_buffer;
priv = jitterbuffer->priv;
if (!gst_rtcp_buffer_get_first_packet (buffer, &packet))
goto invalid_buffer;
/* first packet must be SR or RR or else the validate would have failed */
switch (gst_rtcp_packet_get_type (&packet)) {
case GST_RTCP_TYPE_SR:
gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, NULL, &rtptime,
NULL, NULL);
break;
default:
goto ignore_buffer;
}
GST_DEBUG_OBJECT (jitterbuffer, "received RTCP of SSRC %08x", ssrc);
JBUF_LOCK (priv);
/* convert the RTP timestamp to our extended timestamp, using the same offset
* we used in the jitterbuffer */
ext_rtptime = priv->jbuf->ext_rtptime;
ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
/* get the last values from the jitterbuffer */
rtp_jitter_buffer_get_sync (priv->jbuf, &base_rtptime, &timestamp,
&clock_rate, &last_rtptime);
GST_DEBUG_OBJECT (jitterbuffer, "ext SR %" G_GUINT64_FORMAT ", base %"
G_GUINT64_FORMAT ", clock-rate %" G_GUINT32_FORMAT,
ext_rtptime, base_rtptime, clock_rate);
if (base_rtptime == -1 || clock_rate == -1 || timestamp == -1) {
GST_DEBUG_OBJECT (jitterbuffer, "dropping, no RTP values");
drop = TRUE;
} else {
/* we can't accept anything that happened before we did the last resync */
if (base_rtptime > ext_rtptime) {
GST_DEBUG_OBJECT (jitterbuffer, "dropping, older than base time");
drop = TRUE;
} else {
/* the SR RTP timestamp must be something close to what we last observed
* in the jitterbuffer */
if (ext_rtptime > last_rtptime) {
/* check how far ahead it is to our RTP timestamps */
diff = ext_rtptime - last_rtptime;
/* if bigger than 1 second, we drop it */
if (diff > clock_rate) {
GST_DEBUG_OBJECT (jitterbuffer, "dropping, too far ahead");
drop = TRUE;
}
GST_DEBUG_OBJECT (jitterbuffer, "ext last %" G_GUINT64_FORMAT ", diff %"
G_GUINT64_FORMAT, last_rtptime, diff);
}
}
}
JBUF_UNLOCK (priv);
if (!drop) {
GstStructure *s;
s = gst_structure_new ("application/x-rtp-sync",
"base-rtptime", G_TYPE_UINT64, base_rtptime,
"base-time", G_TYPE_UINT64, timestamp,
"clock-rate", G_TYPE_UINT, clock_rate,
"sr-ext-rtptime", G_TYPE_UINT64, ext_rtptime,
"sr-buffer", GST_TYPE_BUFFER, buffer, NULL);
GST_DEBUG_OBJECT (jitterbuffer, "signaling sync");
g_signal_emit (jitterbuffer,
gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC], 0, s);
gst_structure_free (s);
} else {
GST_DEBUG_OBJECT (jitterbuffer, "dropping RTCP packet");
ret = GST_FLOW_OK;
}
done:
gst_buffer_unref (buffer);
gst_object_unref (jitterbuffer);
return ret;
invalid_buffer:
{
/* this is not fatal but should be filtered earlier */
GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
("Received invalid RTCP payload, dropping"));
ret = GST_FLOW_OK;
goto done;
}
ignore_buffer:
{
GST_DEBUG_OBJECT (jitterbuffer, "ignoring RTCP packet");
ret = GST_FLOW_OK;
goto done;
}
}
static gboolean
gst_rtp_jitter_buffer_query (GstPad * pad, GstQuery * query)
{
GstRtpJitterBuffer *jitterbuffer;
GstRtpJitterBufferPrivate *priv;
gboolean res = FALSE;
jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
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;
}
default:
res = gst_pad_query_default (pad, query);
break;
}
gst_object_unref (jitterbuffer);
return res;
}
static void
gst_rtp_jitter_buffer_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec)
{
GstRtpJitterBuffer *jitterbuffer;
GstRtpJitterBufferPrivate *priv;
jitterbuffer = GST_RTP_JITTER_BUFFER (object);
priv = jitterbuffer->priv;
switch (prop_id) {
case PROP_LATENCY:
{
guint new_latency, old_latency;
new_latency = g_value_get_uint (value);
JBUF_LOCK (priv);
old_latency = priv->latency_ms;
priv->latency_ms = new_latency;
priv->latency_ns = priv->latency_ms * GST_MSECOND;
rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
JBUF_UNLOCK (priv);
/* post message if latency changed, this will inform the parent pipeline
* that a latency reconfiguration is possible/needed. */
if (new_latency != old_latency) {
GST_DEBUG_OBJECT (jitterbuffer, "latency changed to: %" GST_TIME_FORMAT,
GST_TIME_ARGS (new_latency * GST_MSECOND));
gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer),
gst_message_new_latency (GST_OBJECT_CAST (jitterbuffer)));
}
break;
}
case PROP_DROP_ON_LATENCY:
JBUF_LOCK (priv);
priv->drop_on_latency = g_value_get_boolean (value);
JBUF_UNLOCK (priv);
break;
case PROP_TS_OFFSET:
JBUF_LOCK (priv);
priv->ts_offset = g_value_get_int64 (value);
/* FIXME, we don't really have a method for signaling a timestamp
* DISCONT without also making this a data discont. */
/* priv->discont = TRUE; */
JBUF_UNLOCK (priv);
break;
case PROP_DO_LOST:
JBUF_LOCK (priv);
priv->do_lost = g_value_get_boolean (value);
JBUF_UNLOCK (priv);
break;
case PROP_MODE:
JBUF_LOCK (priv);
rtp_jitter_buffer_set_mode (priv->jbuf, g_value_get_enum (value));
JBUF_UNLOCK (priv);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_rtp_jitter_buffer_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec)
{
GstRtpJitterBuffer *jitterbuffer;
GstRtpJitterBufferPrivate *priv;
jitterbuffer = GST_RTP_JITTER_BUFFER (object);
priv = jitterbuffer->priv;
switch (prop_id) {
case PROP_LATENCY:
JBUF_LOCK (priv);
g_value_set_uint (value, priv->latency_ms);
JBUF_UNLOCK (priv);
break;
case PROP_DROP_ON_LATENCY:
JBUF_LOCK (priv);
g_value_set_boolean (value, priv->drop_on_latency);
JBUF_UNLOCK (priv);
break;
case PROP_TS_OFFSET:
JBUF_LOCK (priv);
g_value_set_int64 (value, priv->ts_offset);
JBUF_UNLOCK (priv);
break;
case PROP_DO_LOST:
JBUF_LOCK (priv);
g_value_set_boolean (value, priv->do_lost);
JBUF_UNLOCK (priv);
break;
case PROP_MODE:
JBUF_LOCK (priv);
g_value_set_enum (value, rtp_jitter_buffer_get_mode (priv->jbuf));
JBUF_UNLOCK (priv);
break;
case PROP_PERCENT:
{
gint percent;
JBUF_LOCK (priv);
if (priv->srcresult != GST_FLOW_OK)
percent = 100;
else
percent = rtp_jitter_buffer_get_percent (priv->jbuf);
g_value_set_int (value, percent);
JBUF_UNLOCK (priv);
break;
}
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
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