gstreamer/gst/rtpmanager/gstrtpjitterbuffer.c
Stefan Kost f56adf75ca Add stdlib include (free, atoi, exit).
Original commit message from CVS:
* examples/app/appsrc_ex.c:
* examples/switch/switcher.c:
* ext/neon/gstneonhttpsrc.c:
* ext/timidity/gstwildmidi.c:
* ext/x264/gstx264enc.c:
* gst/mve/mveaudioenc.c: (mve_compress_audio):
* gst/rtpmanager/gstrtpclient.c:
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/spectrum/demo-audiotest.c:
* gst/spectrum/demo-osssrc.c:
* sys/dvb/gstdvbsrc.c:
Add stdlib include (free, atoi, exit).
2007-07-18 07:35:32 +00:00

1238 lines
37 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@fluendo.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
* @short_description: buffer, reorder and remove duplicate RTP packets to
* compensate for network oddities.
*
* <refsect2>
* <para>
* 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 ::latency property. Packets arriving too
* late are considered to be lost packets.
* </para>
* <para>
* This element acts as a live element and so adds ::latency to the pipeline.
* </para>
* <para>
* 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 ::request-pt-map signal. To clear the
* previous pt-map use the ::clear-pt-map signal.
* </para>
* <para>
* This element will automatically be used inside gstrtpbin.
* </para>
* <title>Example pipelines</title>
* <para>
* <programlisting>
* gst-launch rtspsrc location=rtsp://192.168.1.133:8554/mpeg1or2AudioVideoTest ! gstrtpjitterbuffer ! rtpmpvdepay ! mpeg2dec ! xvimagesink
* </programlisting>
* 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.
* </para>
* </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 "async_jitter_queue.h"
GST_DEBUG_CATEGORY (rtpjitterbuffer_debug);
#define GST_CAT_DEFAULT (rtpjitterbuffer_debug)
/* low and high threshold tell the queue when to start and stop buffering */
#define LOW_THRESHOLD 0.2
#define HIGH_THRESHOLD 0.8
/* elementfactory information */
static const GstElementDetails gst_rtp_jitter_buffer_details =
GST_ELEMENT_DETAILS ("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@fluendo.com>");
/* RTPJitterBuffer signals and args */
enum
{
SIGNAL_REQUEST_PT_MAP,
SIGNAL_CLEAR_PT_MAP,
LAST_SIGNAL
};
#define DEFAULT_LATENCY_MS 200
#define DEFAULT_DROP_ON_LATENCY FALSE
enum
{
PROP_0,
PROP_LATENCY,
PROP_DROP_ON_LATENCY
};
struct _GstRTPJitterBufferPrivate
{
GstPad *sinkpad, *srcpad;
AsyncJitterQueue *queue;
/* properties */
guint latency_ms;
gboolean drop_on_latency;
/* the last seqnum we pushed out */
guint32 last_popped_seqnum;
/* the next expected seqnum */
guint32 next_seqnum;
/* clock rate and rtp timestamp offset */
gint32 clock_rate;
gint64 clock_base;
/* when we are shutting down */
GstFlowReturn srcresult;
/* for sync */
GstSegment segment;
GstClockID clock_id;
guint32 waiting_seqnum;
/* 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_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_dispose (GObject * object);
/* element overrides */
static GstStateChangeReturn gst_rtp_jitter_buffer_change_state (GstElement
* element, GstStateChange transition);
/* pad overrides */
static GstCaps *gst_rtp_jitter_buffer_getcaps (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);
/* srcpad overrides */
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 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_set_details (element_class, &gst_rtp_jitter_buffer_details);
}
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->dispose = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_dispose);
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::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::clear-pt-map:
* @buffer: the object which received the signal
*
* Invalidate the clock-rate as obtained with the ::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_STRUCT_OFFSET (GstRTPJitterBufferClass,
clear_pt_map), NULL, NULL, g_cclosure_marshal_VOID__VOID,
G_TYPE_NONE, 0, G_TYPE_NONE);
gstelement_class->change_state = gst_rtp_jitter_buffer_change_state;
klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_clear_pt_map);
GST_DEBUG_CATEGORY_INIT
(rtpjitterbuffer_debug, "rtpjitterbuffer", 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->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
priv->queue = async_jitter_queue_new ();
async_jitter_queue_set_low_threshold (priv->queue, LOW_THRESHOLD);
async_jitter_queue_set_high_threshold (priv->queue, HIGH_THRESHOLD);
priv->waiting_seqnum = -1;
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));
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_dispose (GObject * object)
{
GstRTPJitterBuffer *jitterbuffer;
jitterbuffer = GST_RTP_JITTER_BUFFER (object);
if (jitterbuffer->priv->queue) {
async_jitter_queue_unref (jitterbuffer->priv->queue);
jitterbuffer->priv->queue = NULL;
}
G_OBJECT_CLASS (parent_class)->dispose (object);
}
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. */
priv->clock_rate = -1;
}
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;
}
static gboolean
gst_jitter_buffer_sink_parse_caps (GstRTPJitterBuffer * jitterbuffer,
GstCaps * caps)
{
GstRTPJitterBufferPrivate *priv;
GstStructure *caps_struct;
const GValue *value;
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);
/* gah, clock-base is uint. If we don't have a base, we will use the first
* buffer timestamp as the base time. This will screw up sync but it's better
* than nothing. */
value = gst_structure_get_value (caps_struct, "clock-base");
if (value && G_VALUE_HOLDS_UINT (value)) {
priv->clock_base = g_value_get_uint (value);
GST_DEBUG_OBJECT (jitterbuffer, "got clock-base %" G_GINT64_FORMAT,
priv->clock_base);
} else
priv->clock_base = -1;
/* first expected seqnum */
value = gst_structure_get_value (caps_struct, "seqnum-base");
if (value && G_VALUE_HOLDS_UINT (value)) {
priv->next_seqnum = g_value_get_uint (value);
GST_DEBUG_OBJECT (jitterbuffer, "got seqnum-base %d", priv->next_seqnum);
} else
priv->next_seqnum = -1;
async_jitter_queue_set_max_queue_length (priv->queue,
priv->latency_ms * priv->clock_rate / 1000);
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;
res = gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
/* set same caps on srcpad on success */
if (res)
gst_pad_set_caps (priv->srcpad, caps);
gst_object_unref (jitterbuffer);
return res;
}
static void
free_func (gpointer data, GstRTPJitterBuffer * user_data)
{
if (GST_IS_BUFFER (data))
gst_buffer_unref (GST_BUFFER_CAST (data));
else
gst_event_unref (GST_EVENT_CAST (data));
}
static void
gst_rtp_jitter_buffer_flush_start (GstRTPJitterBuffer * jitterbuffer)
{
GstRTPJitterBufferPrivate *priv;
priv = jitterbuffer->priv;
async_jitter_queue_lock (priv->queue);
/* 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 */
async_jitter_queue_set_flushing_unlocked (jitterbuffer->priv->queue,
(GFunc) free_func, jitterbuffer);
/* 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);
async_jitter_queue_unlock (priv->queue);
}
static void
gst_rtp_jitter_buffer_flush_stop (GstRTPJitterBuffer * jitterbuffer)
{
GstRTPJitterBufferPrivate *priv;
priv = jitterbuffer->priv;
async_jitter_queue_lock (priv->queue);
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->next_seqnum = -1;
priv->clock_rate = -1;
/* allow pops from the src pad task */
async_jitter_queue_unset_flushing_unlocked (jitterbuffer->priv->queue);
async_jitter_queue_unlock (priv->queue);
}
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:
async_jitter_queue_lock (priv->queue);
/* reset negotiated values */
priv->clock_rate = -1;
priv->clock_base = -1;
/* block until we go to PLAYING */
async_jitter_queue_set_blocking_unlocked (jitterbuffer->priv->queue,
TRUE);
async_jitter_queue_unlock (priv->queue);
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
async_jitter_queue_lock (priv->queue);
/* unblock to allow streaming in PLAYING */
async_jitter_queue_set_blocking_unlocked (jitterbuffer->priv->queue,
FALSE);
async_jitter_queue_unlock (priv->queue);
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:
async_jitter_queue_lock (priv->queue);
/* block to stop streaming when PAUSED */
async_jitter_queue_set_blocking_unlocked (jitterbuffer->priv->queue,
TRUE);
async_jitter_queue_unlock (priv->queue);
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;
}
/**
* Performs comparison 'b - a' with check for overflows.
*/
static inline gint
priv_compare_rtp_seq_lt (guint16 a, guint16 b)
{
/* check if diff more than half of the 16bit range */
if (abs (b - a) > (1 << 15)) {
/* one of a/b has wrapped */
return a - b;
} else {
return b - a;
}
}
/**
* gets the seqnum from the buffers and compare them
*/
static gint
compare_rtp_buffers_seq_num (GstBuffer * a, GstBuffer * b)
{
gint ret;
if (GST_IS_BUFFER (a) && GST_IS_BUFFER (b)) {
/* two buffers */
ret = priv_compare_rtp_seq_lt
(gst_rtp_buffer_get_seq (GST_BUFFER_CAST (a)),
gst_rtp_buffer_get_seq (GST_BUFFER_CAST (b)));
} else {
/* one of them is an event, the event always goes before the other element
* so we return -1. */
if (GST_IS_EVENT (a))
ret = -1;
else
ret = 1;
}
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 */
async_jitter_queue_lock (priv->queue);
GST_DEBUG_OBJECT (jitterbuffer, "queuing EOS");
/* check for flushing, we need to discard the event and return FALSE when
* we are flushing */
ret = priv->srcresult == GST_FLOW_OK;
if (ret)
async_jitter_queue_push_unlocked (priv->queue, event);
else
gst_event_unref (event);
async_jitter_queue_unlock (priv->queue);
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_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);
g_signal_emitv (args, gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP], 0,
&ret);
caps = (GstCaps *) g_value_get_boxed (&ret);
if (!caps)
goto no_caps;
res = gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
return res;
/* ERRORS */
no_caps:
{
GST_DEBUG_OBJECT (jitterbuffer, "could not get caps");
return FALSE;
}
}
static GstFlowReturn
gst_rtp_jitter_buffer_chain (GstPad * pad, GstBuffer * buffer)
{
GstRTPJitterBuffer *jitterbuffer;
GstRTPJitterBufferPrivate *priv;
guint16 seqnum;
GstFlowReturn ret;
jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
if (!gst_rtp_buffer_validate (buffer))
goto invalid_buffer;
priv = jitterbuffer->priv;
if (priv->clock_rate == -1) {
guint8 pt;
/* no clock rate given on the caps, try to get one with the signal */
pt = gst_rtp_buffer_get_payload_type (buffer);
gst_rtp_jitter_buffer_get_clock_rate (jitterbuffer, pt);
if (priv->clock_rate == -1)
goto not_negotiated;
}
seqnum = gst_rtp_buffer_get_seq (buffer);
GST_DEBUG_OBJECT (jitterbuffer, "Received packet #%d", seqnum);
async_jitter_queue_lock (priv->queue);
ret = priv->srcresult;
if (ret != GST_FLOW_OK)
goto out_flushing;
/* let's check if this buffer is too late, we cannot accept packets with
* bigger seqnum than the one we already pushed. */
if (priv->last_popped_seqnum != -1) {
if (priv_compare_rtp_seq_lt (priv->last_popped_seqnum, seqnum) < 0)
goto too_late;
}
/* let's drop oldest packet if the queue is already full and drop-on-latency
* is set. */
if (priv->drop_on_latency) {
if (async_jitter_queue_length_ts_units_unlocked (priv->queue) >=
priv->latency_ms * priv->clock_rate / 1000) {
GstBuffer *old_buf;
GST_DEBUG_OBJECT (jitterbuffer, "Queue full, dropping old packet #%d",
seqnum);
old_buf = async_jitter_queue_pop_unlocked (priv->queue);
gst_buffer_unref (old_buf);
}
}
/* 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 (!async_jitter_queue_push_sorted_unlocked (priv->queue, buffer,
(GCompareDataFunc) compare_rtp_buffers_seq_num, NULL))
goto duplicate;
/* let's unschedule and unblock any waiting buffers. We only want to do this
* if there is a currently waiting newer (> seqnum) buffer */
if (priv->clock_id) {
if (priv->waiting_seqnum > seqnum) {
gst_clock_id_unschedule (priv->clock_id);
GST_DEBUG_OBJECT (jitterbuffer, "Unscheduling waiting buffer");
}
}
GST_DEBUG_OBJECT (jitterbuffer, "Pushed packet #%d on queue %d",
seqnum, async_jitter_queue_length_unlocked (priv->queue));
finished:
async_jitter_queue_unlock (priv->queue);
gst_object_unref (jitterbuffer);
return ret;
/* ERRORS */
invalid_buffer:
{
/* this is fatal and should be filtered earlier */
GST_ELEMENT_ERROR (jitterbuffer, STREAM, DECODE, (NULL),
("Received invalid RTP payload"));
gst_buffer_unref (buffer);
gst_object_unref (jitterbuffer);
return GST_FLOW_ERROR;
}
not_negotiated:
{
GST_DEBUG_OBJECT (jitterbuffer, "No clock-rate in caps!");
gst_buffer_unref (buffer);
gst_object_unref (jitterbuffer);
return GST_FLOW_NOT_NEGOTIATED;
}
out_flushing:
{
GST_DEBUG_OBJECT (jitterbuffer, "flushing %s", gst_flow_get_name (ret));
gst_buffer_unref (buffer);
goto finished;
}
too_late:
{
GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d too late as #%d was already"
" popped, dropping", seqnum, priv->last_popped_seqnum);
priv->num_late++;
gst_buffer_unref (buffer);
goto finished;
}
duplicate:
{
GST_DEBUG_OBJECT (jitterbuffer, "Duplicate packet #%d detected, dropping",
seqnum);
priv->num_duplicates++;
gst_buffer_unref (buffer);
goto finished;
}
}
/**
* 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 rtp timestamp of buffer B.
*/
static void
gst_rtp_jitter_buffer_loop (GstRTPJitterBuffer * jitterbuffer)
{
GstRTPJitterBufferPrivate *priv;
gpointer elem;
GstBuffer *outbuf;
GstFlowReturn result;
guint16 seqnum;
guint32 rtp_time;
GstClockTime timestamp;
gint64 running_time;
priv = jitterbuffer->priv;
async_jitter_queue_lock (priv->queue);
again:
GST_DEBUG_OBJECT (jitterbuffer, "Popping item");
/* pop a buffer, we will get NULL if the queue was shut down */
elem = async_jitter_queue_pop_unlocked (priv->queue);
if (!elem)
goto no_elem;
/* special code for events */
if (G_UNLIKELY (GST_IS_EVENT (elem))) {
GstEvent *event = GST_EVENT_CAST (elem);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_EOS:
GST_DEBUG_OBJECT (jitterbuffer, "Popped EOS from queue");
/* we don't expect more data now, makes upstream perform EOS actions */
priv->srcresult = GST_FLOW_UNEXPECTED;
break;
default:
GST_DEBUG_OBJECT (jitterbuffer, "Popped event %s from queue",
GST_EVENT_TYPE_NAME (event));
break;
}
async_jitter_queue_unlock (priv->queue);
/* push event */
gst_pad_push_event (priv->srcpad, event);
return;
}
/* we know it's a buffer now */
outbuf = GST_BUFFER_CAST (elem);
seqnum = gst_rtp_buffer_get_seq (outbuf);
GST_DEBUG_OBJECT (jitterbuffer, "Popped buffer #%d from queue %d",
gst_rtp_buffer_get_seq (outbuf),
async_jitter_queue_length_unlocked (priv->queue));
/* 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 (priv->next_seqnum == -1 || priv->next_seqnum != seqnum) {
GstClockID id;
GstClockTimeDiff jitter;
GstClockReturn ret;
GstClock *clock;
if (priv->next_seqnum != -1) {
/* we expected next_seqnum but received something else, that's a gap */
GST_DEBUG_OBJECT (jitterbuffer,
"Sequence number GAP detected -> %d instead of %d", priv->next_seqnum,
seqnum);
} 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);
}
/* get the max deadline to wait for the missing packets, this is the time
* of the currently popped packet */
rtp_time = gst_rtp_buffer_get_timestamp (outbuf);
GST_DEBUG_OBJECT (jitterbuffer, "rtp_time %u, base %" G_GINT64_FORMAT,
rtp_time, priv->clock_base);
/* if no clock_base was given, take first ts as base */
if (priv->clock_base == -1)
priv->clock_base = rtp_time;
/* take rtp timestamp offset into account, this can wrap around */
rtp_time -= priv->clock_base;
/* bring timestamp to gst time */
timestamp = gst_util_uint64_scale (GST_SECOND, rtp_time, priv->clock_rate);
GST_DEBUG_OBJECT (jitterbuffer,
"rtptime %u, clock-rate %u, timestamp %" GST_TIME_FORMAT, rtp_time,
priv->clock_rate, GST_TIME_ARGS (timestamp));
/* bring to running time */
running_time = gst_segment_to_running_time (&priv->segment, GST_FORMAT_TIME,
timestamp);
/* correct for sync against the gstreamer clock, add latency */
GST_OBJECT_LOCK (jitterbuffer);
clock = GST_ELEMENT_CLOCK (jitterbuffer);
if (!clock) {
GST_OBJECT_UNLOCK (jitterbuffer);
/* let's just push if there is no clock */
goto push_buffer;
}
/* add latency */
running_time += (priv->latency_ms * GST_MSECOND);
GST_DEBUG_OBJECT (jitterbuffer, "sync to running_time %" GST_TIME_FORMAT,
GST_TIME_ARGS (running_time));
/* prepare for sync against clock */
running_time += GST_ELEMENT_CAST (jitterbuffer)->base_time;
/* create an entry for the clock */
id = priv->clock_id = gst_clock_new_single_shot_id (clock, running_time);
priv->waiting_seqnum = seqnum;
GST_OBJECT_UNLOCK (jitterbuffer);
/* release the lock so that the other end can push stuff or unlock */
async_jitter_queue_unlock (priv->queue);
ret = gst_clock_id_wait (id, &jitter);
async_jitter_queue_lock (priv->queue);
/* and free the entry */
gst_clock_id_unref (id);
priv->clock_id = NULL;
priv->waiting_seqnum = -1;
/* 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 (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. Grab it and try to push or sync. */
if (ret == GST_CLOCK_UNSCHEDULED) {
GST_DEBUG_OBJECT (jitterbuffer,
"Wait got unscheduled, will retry to push with new buffer");
/* reinserting popped buffer into queue */
if (!async_jitter_queue_push_sorted_unlocked (priv->queue, outbuf,
(GCompareDataFunc) compare_rtp_buffers_seq_num, NULL)) {
GST_DEBUG_OBJECT (jitterbuffer,
"Duplicate packet #%d detected, dropping", seqnum);
priv->num_duplicates++;
gst_buffer_unref (outbuf);
}
goto again;
}
}
push_buffer:
/* check if we are pushing something unexpected */
if (priv->next_seqnum != -1 && priv->next_seqnum != seqnum) {
gint dropped;
/* calc number of missing packets, careful for wraparounds */
dropped = priv_compare_rtp_seq_lt (priv->next_seqnum, seqnum);
GST_DEBUG_OBJECT (jitterbuffer,
"Pushing DISCONT after dropping %d (%d to %d)", dropped,
priv->next_seqnum, seqnum);
/* update stats */
priv->num_late += dropped;
/* set DISCONT flag */
outbuf = gst_buffer_make_metadata_writable (outbuf);
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
}
/* 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->next_seqnum = (seqnum + 1) & 0xffff;
async_jitter_queue_unlock (priv->queue);
/* push buffer */
GST_DEBUG_OBJECT (jitterbuffer, "Pushing buffer %d", seqnum);
result = gst_pad_push (priv->srcpad, outbuf);
if (result != GST_FLOW_OK)
goto pause;
return;
/* ERRORS */
no_elem:
{
/* store result, we are flushing now */
GST_DEBUG_OBJECT (jitterbuffer, "Pop returned NULL, we're flushing");
priv->srcresult = GST_FLOW_WRONG_STATE;
gst_pad_pause_task (priv->srcpad);
async_jitter_queue_unlock (priv->queue);
return;
}
flushing:
{
GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
gst_buffer_unref (outbuf);
async_jitter_queue_unlock (priv->queue);
return;
}
pause:
{
const gchar *reason = gst_flow_get_name (result);
GST_DEBUG_OBJECT (jitterbuffer, "pausing task, reason %s", reason);
async_jitter_queue_lock (priv->queue);
/* 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);
async_jitter_queue_unlock (priv->queue);
return;
}
}
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;
GstPad *peer;
if ((peer = gst_pad_get_peer (priv->sinkpad))) {
if ((res = gst_pad_query (peer, 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));
min_latency += priv->latency_ms * GST_MSECOND;
max_latency += priv->latency_ms * GST_MSECOND;
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);
}
gst_object_unref (peer);
}
break;
}
default:
res = gst_pad_query_default (pad, query);
break;
}
return res;
}
static void
gst_rtp_jitter_buffer_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec)
{
GstRTPJitterBuffer *jitterbuffer = GST_RTP_JITTER_BUFFER (object);
switch (prop_id) {
case PROP_LATENCY:
{
guint new_latency, old_latency;
/* FIXME, not threadsafe */
new_latency = g_value_get_uint (value);
old_latency = jitterbuffer->priv->latency_ms;
jitterbuffer->priv->latency_ms = new_latency;
if (jitterbuffer->priv->clock_rate != -1) {
async_jitter_queue_set_max_queue_length (jitterbuffer->priv->queue,
gst_util_uint64_scale_int (new_latency,
jitterbuffer->priv->clock_rate, 1000));
}
/* post message if latency changed, this will infor the parent pipeline
* that a latency reconfiguration is possible. */
if (new_latency != old_latency) {
gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer),
gst_message_new_latency (GST_OBJECT_CAST (jitterbuffer)));
}
break;
}
case PROP_DROP_ON_LATENCY:
{
jitterbuffer->priv->drop_on_latency = g_value_get_boolean (value);
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 = GST_RTP_JITTER_BUFFER (object);
switch (prop_id) {
case PROP_LATENCY:
g_value_set_uint (value, jitterbuffer->priv->latency_ms);
break;
case PROP_DROP_ON_LATENCY:
g_value_set_boolean (value, jitterbuffer->priv->drop_on_latency);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}