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9c94f1187c
The scenario where you have a gap in a steady flow of packets of say 10 seconds (500 packets of with duration of 20ms), the jitterbuffer will idle up until it receives the first buffer after the gap, but will then go on to produce 499 lost-events, to "cover up" the gap. Now this is obviously wrong, since the last possible time for the earliest lost-events to be played out has obviously expired, but the fact that the jitterbuffer has a "length", represented with its own latency combined with the total latency downstream, allows for covering up at least some of this gap. So in the case of the "length" being 200ms, while having received packet 500, the jitterbuffer should still create a timeout for packet 491, which will have its time expire at 10,02 seconds, specially since it might actually arrive in time! But obviously, waiting for packet 100, that had its time expire at 2 seconds, (remembering that the current time is 10) is useless... The patch will create one "big" lost-event for the first 490 packets, and then go on to create single ones if they can reach their playout deadline. See https://bugzilla.gnome.org/show_bug.cgi?id=667838
2371 lines
72 KiB
C
2371 lines
72 KiB
C
/*
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* Farsight Voice+Video library
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*
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* Copyright 2007 Collabora Ltd,
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* Copyright 2007 Nokia Corporation
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* @author: Philippe Kalaf <philippe.kalaf@collabora.co.uk>.
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* Copyright 2007 Wim Taymans <wim.taymans@gmail.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*
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*/
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/**
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* SECTION:element-gstrtpjitterbuffer
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*
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* This element reorders and removes duplicate RTP packets as they are received
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* from a network source. It will also wait for missing packets up to a
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* configurable time limit using the #GstRtpJitterBuffer:latency property.
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* Packets arriving too late are considered to be lost packets.
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*
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* This element acts as a live element and so adds #GstRtpJitterBuffer:latency
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* to the pipeline.
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*
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* The element needs the clock-rate of the RTP payload in order to estimate the
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* delay. This information is obtained either from the caps on the sink pad or,
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* when no caps are present, from the #GstRtpJitterBuffer::request-pt-map signal.
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* To clear the previous pt-map use the #GstRtpJitterBuffer::clear-pt-map signal.
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*
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* This element will automatically be used inside gstrtpbin.
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*
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* <refsect2>
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* <title>Example pipelines</title>
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* |[
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* gst-launch-1.0 rtspsrc location=rtsp://192.168.1.133:8554/mpeg1or2AudioVideoTest ! gstrtpjitterbuffer ! rtpmpvdepay ! mpeg2dec ! xvimagesink
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* ]| Connect to a streaming server and decode the MPEG video. The jitterbuffer is
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* inserted into the pipeline to smooth out network jitter and to reorder the
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* out-of-order RTP packets.
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* </refsect2>
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*
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* Last reviewed on 2007-05-28 (0.10.5)
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <stdlib.h>
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#include <string.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include "gstrtpbin-marshal.h"
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#include "gstrtpjitterbuffer.h"
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#include "rtpjitterbuffer.h"
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#include "rtpstats.h"
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#include <gst/glib-compat-private.h>
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GST_DEBUG_CATEGORY (rtpjitterbuffer_debug);
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#define GST_CAT_DEFAULT (rtpjitterbuffer_debug)
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/* RTPJitterBuffer signals and args */
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enum
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{
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SIGNAL_REQUEST_PT_MAP,
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SIGNAL_CLEAR_PT_MAP,
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SIGNAL_HANDLE_SYNC,
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SIGNAL_ON_NPT_STOP,
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SIGNAL_SET_ACTIVE,
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LAST_SIGNAL
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};
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#define DEFAULT_LATENCY_MS 200
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#define DEFAULT_DROP_ON_LATENCY FALSE
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#define DEFAULT_TS_OFFSET 0
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#define DEFAULT_DO_LOST FALSE
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#define DEFAULT_MODE RTP_JITTER_BUFFER_MODE_SLAVE
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#define DEFAULT_PERCENT 0
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enum
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{
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PROP_0,
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PROP_LATENCY,
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PROP_DROP_ON_LATENCY,
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PROP_TS_OFFSET,
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PROP_DO_LOST,
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PROP_MODE,
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PROP_PERCENT,
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PROP_LAST
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};
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#define JBUF_LOCK(priv) (g_mutex_lock (&(priv)->jbuf_lock))
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#define JBUF_LOCK_CHECK(priv,label) G_STMT_START { \
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JBUF_LOCK (priv); \
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if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
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goto label; \
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} G_STMT_END
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#define JBUF_UNLOCK(priv) (g_mutex_unlock (&(priv)->jbuf_lock))
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#define JBUF_WAIT(priv) (g_cond_wait (&(priv)->jbuf_cond, &(priv)->jbuf_lock))
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#define JBUF_WAIT_CHECK(priv,label) G_STMT_START { \
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JBUF_WAIT(priv); \
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if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
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goto label; \
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} G_STMT_END
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#define JBUF_SIGNAL(priv) (g_cond_signal (&(priv)->jbuf_cond))
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struct _GstRtpJitterBufferPrivate
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{
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GstPad *sinkpad, *srcpad;
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GstPad *rtcpsinkpad;
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RTPJitterBuffer *jbuf;
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GMutex jbuf_lock;
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GCond jbuf_cond;
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gboolean waiting;
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gboolean discont;
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gboolean active;
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guint64 out_offset;
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/* properties */
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guint latency_ms;
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guint64 latency_ns;
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gboolean drop_on_latency;
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gint64 ts_offset;
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gboolean do_lost;
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/* the last seqnum we pushed out */
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guint32 last_popped_seqnum;
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/* the next expected seqnum we push */
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guint32 next_seqnum;
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/* last output time */
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GstClockTime last_out_time;
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/* the next expected seqnum we receive */
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guint32 next_in_seqnum;
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/* start and stop ranges */
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GstClockTime npt_start;
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GstClockTime npt_stop;
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guint64 ext_timestamp;
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guint64 last_elapsed;
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guint64 estimated_eos;
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GstClockID eos_id;
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gboolean reached_npt_stop;
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/* state */
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gboolean eos;
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/* clock rate and rtp timestamp offset */
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gint last_pt;
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gint32 clock_rate;
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gint64 clock_base;
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gint64 prev_ts_offset;
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/* when we are shutting down */
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GstFlowReturn srcresult;
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gboolean blocked;
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/* for sync */
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GstSegment segment;
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GstClockID clock_id;
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gboolean unscheduled;
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/* the latency of the upstream peer, we have to take this into account when
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* synchronizing the buffers. */
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GstClockTime peer_latency;
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/* some accounting */
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guint64 num_late;
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guint64 num_duplicates;
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};
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#define GST_RTP_JITTER_BUFFER_GET_PRIVATE(o) \
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(G_TYPE_INSTANCE_GET_PRIVATE ((o), GST_TYPE_RTP_JITTER_BUFFER, \
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GstRtpJitterBufferPrivate))
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static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_template =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("application/x-rtp, "
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"clock-rate = (int) [ 1, 2147483647 ]"
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/* "payload = (int) , "
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* "encoding-name = (string) "
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*/ )
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);
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static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_rtcp_template =
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GST_STATIC_PAD_TEMPLATE ("sink_rtcp",
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GST_PAD_SINK,
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GST_PAD_REQUEST,
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GST_STATIC_CAPS ("application/x-rtcp")
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);
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static GstStaticPadTemplate gst_rtp_jitter_buffer_src_template =
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GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("application/x-rtp"
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/* "payload = (int) , "
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* "clock-rate = (int) , "
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* "encoding-name = (string) "
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*/ )
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);
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static guint gst_rtp_jitter_buffer_signals[LAST_SIGNAL] = { 0 };
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#define gst_rtp_jitter_buffer_parent_class parent_class
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G_DEFINE_TYPE (GstRtpJitterBuffer, gst_rtp_jitter_buffer, GST_TYPE_ELEMENT);
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/* object overrides */
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static void gst_rtp_jitter_buffer_set_property (GObject * object,
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guint prop_id, const GValue * value, GParamSpec * pspec);
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static void gst_rtp_jitter_buffer_get_property (GObject * object,
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guint prop_id, GValue * value, GParamSpec * pspec);
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static void gst_rtp_jitter_buffer_finalize (GObject * object);
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/* element overrides */
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static GstStateChangeReturn gst_rtp_jitter_buffer_change_state (GstElement
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* element, GstStateChange transition);
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static GstPad *gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
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GstPadTemplate * templ, const gchar * name, const GstCaps * filter);
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static void gst_rtp_jitter_buffer_release_pad (GstElement * element,
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GstPad * pad);
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static GstClock *gst_rtp_jitter_buffer_provide_clock (GstElement * element);
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/* pad overrides */
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static GstCaps *gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter);
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static GstIterator *gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad,
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GstObject * parent);
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/* sinkpad overrides */
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static gboolean gst_rtp_jitter_buffer_sink_event (GstPad * pad,
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GstObject * parent, GstEvent * event);
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static GstFlowReturn gst_rtp_jitter_buffer_chain (GstPad * pad,
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GstObject * parent, GstBuffer * buffer);
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static gboolean gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad,
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GstObject * parent, GstEvent * event);
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static GstFlowReturn gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad,
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GstObject * parent, GstBuffer * buffer);
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static gboolean gst_rtp_jitter_buffer_sink_query (GstPad * pad,
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GstObject * parent, GstQuery * query);
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/* srcpad overrides */
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static gboolean gst_rtp_jitter_buffer_src_event (GstPad * pad,
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GstObject * parent, GstEvent * event);
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static gboolean gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad,
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GstObject * parent, GstPadMode mode, gboolean active);
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static void gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer);
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static gboolean gst_rtp_jitter_buffer_src_query (GstPad * pad,
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GstObject * parent, GstQuery * query);
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static void
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gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer);
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static GstClockTime
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gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jitterbuffer,
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gboolean active, guint64 base_time);
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static void
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gst_rtp_jitter_buffer_class_init (GstRtpJitterBufferClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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g_type_class_add_private (klass, sizeof (GstRtpJitterBufferPrivate));
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gobject_class->finalize = gst_rtp_jitter_buffer_finalize;
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gobject_class->set_property = gst_rtp_jitter_buffer_set_property;
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gobject_class->get_property = gst_rtp_jitter_buffer_get_property;
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/**
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* GstRtpJitterBuffer::latency:
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*
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* The maximum latency of the jitterbuffer. Packets will be kept in the buffer
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* for at most this time.
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*/
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g_object_class_install_property (gobject_class, PROP_LATENCY,
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g_param_spec_uint ("latency", "Buffer latency in ms",
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"Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstRtpJitterBuffer::drop-on-latency:
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*
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* Drop oldest buffers when the queue is completely filled.
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*/
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g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
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g_param_spec_boolean ("drop-on-latency",
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"Drop buffers when maximum latency is reached",
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"Tells the jitterbuffer to never exceed the given latency in size",
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DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstRtpJitterBuffer::ts-offset:
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*
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* Adjust GStreamer output buffer timestamps in the jitterbuffer with offset.
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* This is mainly used to ensure interstream synchronisation.
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*/
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g_object_class_install_property (gobject_class, PROP_TS_OFFSET,
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g_param_spec_int64 ("ts-offset", "Timestamp Offset",
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"Adjust buffer timestamps with offset in nanoseconds", G_MININT64,
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G_MAXINT64, DEFAULT_TS_OFFSET,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstRtpJitterBuffer::do-lost:
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*
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* Send out a GstRTPPacketLost event downstream when a packet is considered
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* lost.
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*/
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g_object_class_install_property (gobject_class, PROP_DO_LOST,
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g_param_spec_boolean ("do-lost", "Do Lost",
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"Send an event downstream when a packet is lost", DEFAULT_DO_LOST,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstRtpJitterBuffer::mode:
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*
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* Control the buffering and timestamping mode used by the jitterbuffer.
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*/
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g_object_class_install_property (gobject_class, PROP_MODE,
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g_param_spec_enum ("mode", "Mode",
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"Control the buffering algorithm in use", RTP_TYPE_JITTER_BUFFER_MODE,
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DEFAULT_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstRtpJitterBuffer::percent:
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*
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* The percent of the jitterbuffer that is filled.
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*
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* Since: 0.10.19
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*/
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g_object_class_install_property (gobject_class, PROP_PERCENT,
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g_param_spec_int ("percent", "percent",
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"The buffer filled percent", 0, 100,
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0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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/**
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* GstRtpJitterBuffer::request-pt-map:
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* @buffer: the object which received the signal
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* @pt: the pt
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*
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* Request the payload type as #GstCaps for @pt.
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*/
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gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP] =
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g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
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request_pt_map), NULL, NULL, gst_rtp_bin_marshal_BOXED__UINT,
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GST_TYPE_CAPS, 1, G_TYPE_UINT);
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/**
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* GstRtpJitterBuffer::handle-sync:
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* @buffer: the object which received the signal
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* @struct: a GstStructure containing sync values.
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*
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* Be notified of new sync values.
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*/
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gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC] =
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g_signal_new ("handle-sync", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
|
|
handle_sync), NULL, NULL, g_cclosure_marshal_VOID__BOXED,
|
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G_TYPE_NONE, 1, GST_TYPE_STRUCTURE | G_SIGNAL_TYPE_STATIC_SCOPE);
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|
|
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/**
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* GstRtpJitterBuffer::on-npt-stop
|
|
* @buffer: the object which received the signal
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*
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* Signal that the jitterbufer has pushed the RTP packet that corresponds to
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* the npt-stop position.
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*/
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gst_rtp_jitter_buffer_signals[SIGNAL_ON_NPT_STOP] =
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g_signal_new ("on-npt-stop", G_TYPE_FROM_CLASS (klass),
|
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G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
|
|
on_npt_stop), NULL, NULL, g_cclosure_marshal_VOID__VOID,
|
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G_TYPE_NONE, 0, G_TYPE_NONE);
|
|
|
|
/**
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|
* GstRtpJitterBuffer::clear-pt-map:
|
|
* @buffer: the object which received the signal
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*
|
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* Invalidate the clock-rate as obtained with the
|
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* #GstRtpJitterBuffer::request-pt-map signal.
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|
*/
|
|
gst_rtp_jitter_buffer_signals[SIGNAL_CLEAR_PT_MAP] =
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g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
|
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G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
|
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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.
|
|
*
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|
* Returns: the time of the last pushed packet.
|
|
*
|
|
* Since: 0.10.19
|
|
*/
|
|
gst_rtp_jitter_buffer_signals[SIGNAL_SET_ACTIVE] =
|
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g_signal_new ("set-active", G_TYPE_FROM_CLASS (klass),
|
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G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
|
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G_STRUCT_OFFSET (GstRtpJitterBufferClass, set_active), NULL, NULL,
|
|
gst_rtp_bin_marshal_UINT64__BOOL_UINT64, G_TYPE_UINT64, 2, G_TYPE_BOOLEAN,
|
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G_TYPE_UINT64);
|
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|
|
gstelement_class->change_state =
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GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_change_state);
|
|
gstelement_class->request_new_pad =
|
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GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_request_new_pad);
|
|
gstelement_class->release_pad =
|
|
GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_release_pad);
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|
gstelement_class->provide_clock =
|
|
GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_provide_clock);
|
|
|
|
gst_element_class_add_pad_template (gstelement_class,
|
|
gst_static_pad_template_get (&gst_rtp_jitter_buffer_src_template));
|
|
gst_element_class_add_pad_template (gstelement_class,
|
|
gst_static_pad_template_get (&gst_rtp_jitter_buffer_sink_template));
|
|
gst_element_class_add_pad_template (gstelement_class,
|
|
gst_static_pad_template_get (&gst_rtp_jitter_buffer_sink_rtcp_template));
|
|
|
|
gst_element_class_set_static_metadata (gstelement_class,
|
|
"RTP packet jitter-buffer", "Filter/Network/RTP",
|
|
"A buffer that deals with network jitter and other transmission faults",
|
|
"Philippe Kalaf <philippe.kalaf@collabora.co.uk>, "
|
|
"Wim Taymans <wim.taymans@gmail.com>");
|
|
|
|
klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_clear_pt_map);
|
|
klass->set_active = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_set_active);
|
|
|
|
GST_DEBUG_CATEGORY_INIT
|
|
(rtpjitterbuffer_debug, "gstrtpjitterbuffer", 0, "RTP Jitter Buffer");
|
|
}
|
|
|
|
static void
|
|
gst_rtp_jitter_buffer_init (GstRtpJitterBuffer * jitterbuffer)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv;
|
|
|
|
priv = GST_RTP_JITTER_BUFFER_GET_PRIVATE (jitterbuffer);
|
|
jitterbuffer->priv = priv;
|
|
|
|
priv->latency_ms = DEFAULT_LATENCY_MS;
|
|
priv->latency_ns = priv->latency_ms * GST_MSECOND;
|
|
priv->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
|
|
priv->do_lost = DEFAULT_DO_LOST;
|
|
|
|
priv->jbuf = rtp_jitter_buffer_new ();
|
|
g_mutex_init (&priv->jbuf_lock);
|
|
g_cond_init (&priv->jbuf_cond);
|
|
|
|
/* reset skew detection initialy */
|
|
rtp_jitter_buffer_reset_skew (priv->jbuf);
|
|
rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
|
|
rtp_jitter_buffer_set_buffering (priv->jbuf, FALSE);
|
|
priv->active = TRUE;
|
|
|
|
priv->srcpad =
|
|
gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_src_template,
|
|
"src");
|
|
|
|
gst_pad_set_activatemode_function (priv->srcpad,
|
|
GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_activate_mode));
|
|
gst_pad_set_query_function (priv->srcpad,
|
|
GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_query));
|
|
gst_pad_set_event_function (priv->srcpad,
|
|
GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_event));
|
|
|
|
priv->sinkpad =
|
|
gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_sink_template,
|
|
"sink");
|
|
|
|
gst_pad_set_chain_function (priv->sinkpad,
|
|
GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_chain));
|
|
gst_pad_set_event_function (priv->sinkpad,
|
|
GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_event));
|
|
gst_pad_set_query_function (priv->sinkpad,
|
|
GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_query));
|
|
|
|
gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->srcpad);
|
|
gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->sinkpad);
|
|
|
|
GST_OBJECT_FLAG_SET (jitterbuffer, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
|
|
}
|
|
|
|
static void
|
|
gst_rtp_jitter_buffer_finalize (GObject * object)
|
|
{
|
|
GstRtpJitterBuffer *jitterbuffer;
|
|
|
|
jitterbuffer = GST_RTP_JITTER_BUFFER (object);
|
|
|
|
g_mutex_clear (&jitterbuffer->priv->jbuf_lock);
|
|
g_cond_clear (&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, GstObject * parent)
|
|
{
|
|
GstRtpJitterBuffer *jitterbuffer;
|
|
GstPad *otherpad = NULL;
|
|
GstIterator *it;
|
|
GValue val = { 0, };
|
|
|
|
jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
|
|
|
|
if (pad == jitterbuffer->priv->sinkpad) {
|
|
otherpad = jitterbuffer->priv->srcpad;
|
|
} else if (pad == jitterbuffer->priv->srcpad) {
|
|
otherpad = jitterbuffer->priv->sinkpad;
|
|
} else if (pad == jitterbuffer->priv->rtcpsinkpad) {
|
|
otherpad = NULL;
|
|
}
|
|
|
|
g_value_init (&val, GST_TYPE_PAD);
|
|
g_value_set_object (&val, otherpad);
|
|
it = gst_iterator_new_single (GST_TYPE_PAD, &val);
|
|
g_value_unset (&val);
|
|
|
|
return it;
|
|
}
|
|
|
|
static GstPad *
|
|
create_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv;
|
|
|
|
priv = jitterbuffer->priv;
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "creating RTCP sink pad");
|
|
|
|
priv->rtcpsinkpad =
|
|
gst_pad_new_from_static_template
|
|
(&gst_rtp_jitter_buffer_sink_rtcp_template, "sink_rtcp");
|
|
gst_pad_set_chain_function (priv->rtcpsinkpad,
|
|
gst_rtp_jitter_buffer_chain_rtcp);
|
|
gst_pad_set_event_function (priv->rtcpsinkpad,
|
|
(GstPadEventFunction) gst_rtp_jitter_buffer_sink_rtcp_event);
|
|
gst_pad_set_iterate_internal_links_function (priv->rtcpsinkpad,
|
|
gst_rtp_jitter_buffer_iterate_internal_links);
|
|
gst_pad_set_active (priv->rtcpsinkpad, TRUE);
|
|
gst_element_add_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
|
|
|
|
return priv->rtcpsinkpad;
|
|
}
|
|
|
|
static void
|
|
remove_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv;
|
|
|
|
priv = jitterbuffer->priv;
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "removing RTCP sink pad");
|
|
|
|
gst_pad_set_active (priv->rtcpsinkpad, FALSE);
|
|
|
|
gst_element_remove_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
|
|
priv->rtcpsinkpad = NULL;
|
|
}
|
|
|
|
static GstPad *
|
|
gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
|
|
GstPadTemplate * templ, const gchar * name, const GstCaps * filter)
|
|
{
|
|
GstRtpJitterBuffer *jitterbuffer;
|
|
GstElementClass *klass;
|
|
GstPad *result;
|
|
GstRtpJitterBufferPrivate *priv;
|
|
|
|
g_return_val_if_fail (templ != NULL, NULL);
|
|
g_return_val_if_fail (GST_IS_RTP_JITTER_BUFFER (element), NULL);
|
|
|
|
jitterbuffer = GST_RTP_JITTER_BUFFER (element);
|
|
priv = jitterbuffer->priv;
|
|
klass = GST_ELEMENT_GET_CLASS (element);
|
|
|
|
GST_DEBUG_OBJECT (element, "requesting pad %s", GST_STR_NULL (name));
|
|
|
|
/* figure out the template */
|
|
if (templ == gst_element_class_get_pad_template (klass, "sink_rtcp")) {
|
|
if (priv->rtcpsinkpad != NULL)
|
|
goto exists;
|
|
|
|
result = create_rtcp_sink (jitterbuffer);
|
|
} else
|
|
goto wrong_template;
|
|
|
|
return result;
|
|
|
|
/* ERRORS */
|
|
wrong_template:
|
|
{
|
|
g_warning ("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 GstClock *
|
|
gst_rtp_jitter_buffer_provide_clock (GstElement * element)
|
|
{
|
|
return gst_system_clock_obtain ();
|
|
}
|
|
|
|
static void
|
|
gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv;
|
|
|
|
priv = jitterbuffer->priv;
|
|
|
|
/* this will trigger a new pt-map request signal, FIXME, do something better. */
|
|
|
|
JBUF_LOCK (priv);
|
|
priv->clock_rate = -1;
|
|
/* do not clear current content, but refresh state for new arrival */
|
|
GST_DEBUG_OBJECT (jitterbuffer, "reset jitterbuffer");
|
|
rtp_jitter_buffer_reset_skew (priv->jbuf);
|
|
priv->last_popped_seqnum = -1;
|
|
priv->next_seqnum = -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, GstCaps * filter)
|
|
{
|
|
GstRtpJitterBuffer *jitterbuffer;
|
|
GstRtpJitterBufferPrivate *priv;
|
|
GstPad *other;
|
|
GstCaps *caps;
|
|
GstCaps *templ;
|
|
|
|
jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
|
|
priv = jitterbuffer->priv;
|
|
|
|
other = (pad == priv->srcpad ? priv->sinkpad : priv->srcpad);
|
|
|
|
caps = gst_pad_peer_query_caps (other, filter);
|
|
|
|
templ = gst_pad_get_pad_template_caps (pad);
|
|
if (caps == NULL) {
|
|
GST_DEBUG_OBJECT (jitterbuffer, "use template");
|
|
caps = templ;
|
|
} else {
|
|
GstCaps *intersect;
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "intersect with template");
|
|
|
|
intersect = gst_caps_intersect (caps, templ);
|
|
gst_caps_unref (caps);
|
|
gst_caps_unref (templ);
|
|
|
|
caps = intersect;
|
|
}
|
|
gst_object_unref (jitterbuffer);
|
|
|
|
return caps;
|
|
}
|
|
|
|
/*
|
|
* Must be called with JBUF_LOCK held
|
|
*/
|
|
|
|
static gboolean
|
|
gst_jitter_buffer_sink_parse_caps (GstRtpJitterBuffer * jitterbuffer,
|
|
GstCaps * caps)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv;
|
|
GstStructure *caps_struct;
|
|
guint val;
|
|
GstClockTime tval;
|
|
|
|
priv = jitterbuffer->priv;
|
|
|
|
/* first parse the caps */
|
|
caps_struct = gst_caps_get_structure (caps, 0);
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "got caps");
|
|
|
|
/* we need a clock-rate to convert the rtp timestamps to GStreamer time and to
|
|
* measure the amount of data in the buffer */
|
|
if (!gst_structure_get_int (caps_struct, "clock-rate", &priv->clock_rate))
|
|
goto error;
|
|
|
|
if (priv->clock_rate <= 0)
|
|
goto wrong_rate;
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "got clock-rate %d", priv->clock_rate);
|
|
|
|
/* The clock base is the RTP timestamp corrsponding to the npt-start value. We
|
|
* can use this to track the amount of time elapsed on the sender. */
|
|
if (gst_structure_get_uint (caps_struct, "clock-base", &val))
|
|
priv->clock_base = val;
|
|
else
|
|
priv->clock_base = -1;
|
|
|
|
priv->ext_timestamp = priv->clock_base;
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "got clock-base %" G_GINT64_FORMAT,
|
|
priv->clock_base);
|
|
|
|
if (gst_structure_get_uint (caps_struct, "seqnum-base", &val)) {
|
|
/* first expected seqnum, only update when we didn't have a previous base. */
|
|
if (priv->next_in_seqnum == -1)
|
|
priv->next_in_seqnum = val;
|
|
if (priv->next_seqnum == -1)
|
|
priv->next_seqnum = val;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "got seqnum-base %d", priv->next_in_seqnum);
|
|
|
|
/* the start and stop times. The seqnum-base corresponds to the start time. We
|
|
* will keep track of the seqnums on the output and when we reach the one
|
|
* corresponding to npt-stop, we emit the npt-stop-reached signal */
|
|
if (gst_structure_get_clock_time (caps_struct, "npt-start", &tval))
|
|
priv->npt_start = tval;
|
|
else
|
|
priv->npt_start = 0;
|
|
|
|
if (gst_structure_get_clock_time (caps_struct, "npt-stop", &tval))
|
|
priv->npt_stop = tval;
|
|
else
|
|
priv->npt_stop = -1;
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer,
|
|
"npt start/stop: %" GST_TIME_FORMAT "-%" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (priv->npt_start), GST_TIME_ARGS (priv->npt_stop));
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
error:
|
|
{
|
|
GST_DEBUG_OBJECT (jitterbuffer, "No clock-rate in caps!");
|
|
return FALSE;
|
|
}
|
|
wrong_rate:
|
|
{
|
|
GST_DEBUG_OBJECT (jitterbuffer, "Invalid clock-rate %d", priv->clock_rate);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtp_jitter_buffer_flush_start (GstRtpJitterBuffer * jitterbuffer)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv;
|
|
|
|
priv = jitterbuffer->priv;
|
|
|
|
JBUF_LOCK (priv);
|
|
/* mark ourselves as flushing */
|
|
priv->srcresult = GST_FLOW_FLUSHING;
|
|
GST_DEBUG_OBJECT (jitterbuffer, "Disabling pop on queue");
|
|
/* this unblocks any waiting pops on the src pad task */
|
|
JBUF_SIGNAL (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_mode (GstPad * pad, GstObject * parent,
|
|
GstPadMode mode, gboolean active)
|
|
{
|
|
gboolean result;
|
|
GstRtpJitterBuffer *jitterbuffer = NULL;
|
|
|
|
jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
|
|
|
|
switch (mode) {
|
|
case GST_PAD_MODE_PUSH:
|
|
if (active) {
|
|
/* allow data processing */
|
|
gst_rtp_jitter_buffer_flush_stop (jitterbuffer);
|
|
|
|
/* start pushing out buffers */
|
|
GST_DEBUG_OBJECT (jitterbuffer, "Starting task on srcpad");
|
|
result = gst_pad_start_task (jitterbuffer->priv->srcpad,
|
|
(GstTaskFunction) gst_rtp_jitter_buffer_loop, jitterbuffer, NULL);
|
|
} else {
|
|
/* make sure all data processing stops ASAP */
|
|
gst_rtp_jitter_buffer_flush_start (jitterbuffer);
|
|
|
|
/* NOTE this will hardlock if the state change is called from the src pad
|
|
* task thread because we will _join() the thread. */
|
|
GST_DEBUG_OBJECT (jitterbuffer, "Stopping task on srcpad");
|
|
result = gst_pad_stop_task (pad);
|
|
}
|
|
break;
|
|
default:
|
|
result = FALSE;
|
|
break;
|
|
}
|
|
return result;
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_rtp_jitter_buffer_change_state (GstElement * element,
|
|
GstStateChange transition)
|
|
{
|
|
GstRtpJitterBuffer *jitterbuffer;
|
|
GstRtpJitterBufferPrivate *priv;
|
|
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
|
|
|
|
jitterbuffer = GST_RTP_JITTER_BUFFER (element);
|
|
priv = jitterbuffer->priv;
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_NULL_TO_READY:
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
JBUF_LOCK (priv);
|
|
/* reset negotiated values */
|
|
priv->clock_rate = -1;
|
|
priv->clock_base = -1;
|
|
priv->peer_latency = 0;
|
|
priv->last_pt = -1;
|
|
/* block until we go to PLAYING */
|
|
priv->blocked = TRUE;
|
|
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, GstObject * parent,
|
|
GstEvent * event)
|
|
{
|
|
gboolean ret = TRUE;
|
|
GstRtpJitterBuffer *jitterbuffer;
|
|
GstRtpJitterBufferPrivate *priv;
|
|
|
|
jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
|
|
priv = jitterbuffer->priv;
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_LATENCY:
|
|
{
|
|
GstClockTime latency;
|
|
|
|
gst_event_parse_latency (event, &latency);
|
|
|
|
JBUF_LOCK (priv);
|
|
/* adjust the overall buffer delay to the total pipeline latency in
|
|
* buffering mode because if downstream consumes too fast (because of
|
|
* large latency or queues, we would start rebuffering again. */
|
|
if (rtp_jitter_buffer_get_mode (priv->jbuf) ==
|
|
RTP_JITTER_BUFFER_MODE_BUFFER) {
|
|
rtp_jitter_buffer_set_delay (priv->jbuf, latency);
|
|
}
|
|
JBUF_UNLOCK (priv);
|
|
|
|
ret = gst_pad_push_event (priv->sinkpad, event);
|
|
break;
|
|
}
|
|
default:
|
|
ret = gst_pad_push_event (priv->sinkpad, event);
|
|
break;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_jitter_buffer_sink_event (GstPad * pad, GstObject * parent,
|
|
GstEvent * event)
|
|
{
|
|
gboolean ret = TRUE;
|
|
GstRtpJitterBuffer *jitterbuffer;
|
|
GstRtpJitterBufferPrivate *priv;
|
|
|
|
jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
|
|
priv = jitterbuffer->priv;
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_CAPS:
|
|
{
|
|
GstCaps *caps;
|
|
|
|
gst_event_parse_caps (event, &caps);
|
|
|
|
JBUF_LOCK (priv);
|
|
ret = gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
|
|
JBUF_UNLOCK (priv);
|
|
|
|
/* set same caps on srcpad on success */
|
|
if (ret)
|
|
ret = gst_pad_push_event (priv->srcpad, event);
|
|
else
|
|
gst_event_unref (event);
|
|
break;
|
|
}
|
|
case GST_EVENT_SEGMENT:
|
|
{
|
|
gst_event_copy_segment (event, &priv->segment);
|
|
|
|
/* we need time for now */
|
|
if (priv->segment.format != GST_FORMAT_TIME)
|
|
goto newseg_wrong_format;
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer,
|
|
"newsegment: %" GST_SEGMENT_FORMAT, &priv->segment);
|
|
|
|
/* 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_mode (priv->srcpad, parent,
|
|
GST_PAD_MODE_PUSH, 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_INFO_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:
|
|
|
|
return ret;
|
|
|
|
/* ERRORS */
|
|
newseg_wrong_format:
|
|
{
|
|
GST_DEBUG_OBJECT (jitterbuffer, "received non TIME newsegment");
|
|
ret = FALSE;
|
|
gst_event_unref (event);
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad, GstObject * parent,
|
|
GstEvent * event)
|
|
{
|
|
gboolean ret = TRUE;
|
|
GstRtpJitterBuffer *jitterbuffer;
|
|
|
|
jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_FLUSH_START:
|
|
gst_event_unref (event);
|
|
break;
|
|
case GST_EVENT_FLUSH_STOP:
|
|
gst_event_unref (event);
|
|
break;
|
|
default:
|
|
ret = gst_pad_event_default (pad, parent, event);
|
|
break;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
/*
|
|
* Must be called with JBUF_LOCK held, will release the LOCK when emiting the
|
|
* signal. The function returns GST_FLOW_ERROR when a parsing error happened and
|
|
* GST_FLOW_FLUSHING when the element is shutting down. On success
|
|
* GST_FLOW_OK is returned.
|
|
*/
|
|
static GstFlowReturn
|
|
gst_rtp_jitter_buffer_get_clock_rate (GstRtpJitterBuffer * jitterbuffer,
|
|
guint8 pt)
|
|
{
|
|
GValue ret = { 0 };
|
|
GValue args[2] = { {0}, {0} };
|
|
GstCaps *caps;
|
|
gboolean res;
|
|
|
|
g_value_init (&args[0], GST_TYPE_ELEMENT);
|
|
g_value_set_object (&args[0], jitterbuffer);
|
|
g_value_init (&args[1], G_TYPE_UINT);
|
|
g_value_set_uint (&args[1], pt);
|
|
|
|
g_value_init (&ret, GST_TYPE_CAPS);
|
|
g_value_set_boxed (&ret, NULL);
|
|
|
|
JBUF_UNLOCK (jitterbuffer->priv);
|
|
g_signal_emitv (args, gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP], 0,
|
|
&ret);
|
|
JBUF_LOCK_CHECK (jitterbuffer->priv, out_flushing);
|
|
|
|
g_value_unset (&args[0]);
|
|
g_value_unset (&args[1]);
|
|
caps = (GstCaps *) g_value_dup_boxed (&ret);
|
|
g_value_unset (&ret);
|
|
if (!caps)
|
|
goto no_caps;
|
|
|
|
res = gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
|
|
gst_caps_unref (caps);
|
|
|
|
if (G_UNLIKELY (!res))
|
|
goto parse_failed;
|
|
|
|
return GST_FLOW_OK;
|
|
|
|
/* ERRORS */
|
|
no_caps:
|
|
{
|
|
GST_DEBUG_OBJECT (jitterbuffer, "could not get caps");
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
out_flushing:
|
|
{
|
|
GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
|
|
return GST_FLOW_FLUSHING;
|
|
}
|
|
parse_failed:
|
|
{
|
|
GST_DEBUG_OBJECT (jitterbuffer, "parse failed");
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
}
|
|
|
|
/* call with jbuf lock held */
|
|
static void
|
|
check_buffering_percent (GstRtpJitterBuffer * jitterbuffer, gint * percent)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
|
|
|
|
/* too short a stream, or too close to EOS will never really fill buffer */
|
|
if (*percent != -1 && priv->npt_stop != -1 &&
|
|
priv->npt_stop - priv->npt_start <=
|
|
rtp_jitter_buffer_get_delay (priv->jbuf)) {
|
|
GST_DEBUG_OBJECT (jitterbuffer, "short stream; faking full buffer");
|
|
rtp_jitter_buffer_set_buffering (priv->jbuf, FALSE);
|
|
*percent = 100;
|
|
}
|
|
}
|
|
|
|
static void
|
|
post_buffering_percent (GstRtpJitterBuffer * jitterbuffer, gint percent)
|
|
{
|
|
GstMessage *message;
|
|
|
|
/* Post a buffering message */
|
|
message = gst_message_new_buffering (GST_OBJECT_CAST (jitterbuffer), percent);
|
|
gst_message_set_buffering_stats (message, GST_BUFFERING_LIVE, -1, -1, -1);
|
|
|
|
gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), message);
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_rtp_jitter_buffer_chain (GstPad * pad, GstObject * parent,
|
|
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;
|
|
GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
|
|
|
|
jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
|
|
|
|
priv = jitterbuffer->priv;
|
|
|
|
if (G_UNLIKELY (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp)))
|
|
goto invalid_buffer;
|
|
|
|
pt = gst_rtp_buffer_get_payload_type (&rtp);
|
|
seqnum = gst_rtp_buffer_get_seq (&rtp);
|
|
gst_rtp_buffer_unmap (&rtp);
|
|
|
|
/* 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);
|
|
|
|
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_pad_get_current_caps (pad))) {
|
|
gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
|
|
gst_caps_unref (caps);
|
|
}
|
|
}
|
|
|
|
if (G_UNLIKELY (priv->clock_rate == -1)) {
|
|
/* no clock rate given on the caps, try to get one with the signal */
|
|
if (gst_rtp_jitter_buffer_get_clock_rate (jitterbuffer,
|
|
pt) == GST_FLOW_FLUSHING)
|
|
goto out_flushing;
|
|
|
|
if (G_UNLIKELY (priv->clock_rate == -1))
|
|
goto no_clock_rate;
|
|
}
|
|
|
|
/* don't accept more data on EOS */
|
|
if (G_UNLIKELY (priv->eos))
|
|
goto have_eos;
|
|
|
|
/* 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 %p",
|
|
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_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);
|
|
|
|
check_buffering_percent (jitterbuffer, &percent);
|
|
|
|
finished:
|
|
JBUF_UNLOCK (priv);
|
|
|
|
if (percent != -1)
|
|
post_buffering_percent (jitterbuffer, percent);
|
|
|
|
return ret;
|
|
|
|
/* ERRORS */
|
|
invalid_buffer:
|
|
{
|
|
/* this is not fatal but should be filtered earlier */
|
|
GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
|
|
("Received invalid RTP payload, dropping"));
|
|
gst_buffer_unref (buffer);
|
|
return GST_FLOW_OK;
|
|
}
|
|
no_clock_rate:
|
|
{
|
|
GST_WARNING_OBJECT (jitterbuffer,
|
|
"No clock-rate in caps!, dropping buffer");
|
|
gst_buffer_unref (buffer);
|
|
goto finished;
|
|
}
|
|
out_flushing:
|
|
{
|
|
ret = priv->srcresult;
|
|
GST_DEBUG_OBJECT (jitterbuffer, "flushing %s", gst_flow_get_name (ret));
|
|
gst_buffer_unref (buffer);
|
|
goto finished;
|
|
}
|
|
have_eos:
|
|
{
|
|
ret = GST_FLOW_EOS;
|
|
GST_WARNING_OBJECT (jitterbuffer, "we are EOS, refusing buffer");
|
|
gst_buffer_unref (buffer);
|
|
goto finished;
|
|
}
|
|
too_late:
|
|
{
|
|
GST_WARNING_OBJECT (jitterbuffer, "Packet #%d too late as #%d was already"
|
|
" popped, dropping", seqnum, priv->last_popped_seqnum);
|
|
priv->num_late++;
|
|
gst_buffer_unref (buffer);
|
|
goto finished;
|
|
}
|
|
duplicate:
|
|
{
|
|
GST_WARNING_OBJECT (jitterbuffer, "Duplicate packet #%d detected, dropping",
|
|
seqnum);
|
|
priv->num_duplicates++;
|
|
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_INFO_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;
|
|
}
|
|
}
|
|
|
|
static GstClockTime
|
|
compute_elapsed (GstRtpJitterBuffer * jitterbuffer, GstBuffer * outbuf)
|
|
{
|
|
guint64 ext_time, elapsed;
|
|
guint32 rtp_time;
|
|
GstRtpJitterBufferPrivate *priv;
|
|
GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
|
|
|
|
priv = jitterbuffer->priv;
|
|
gst_rtp_buffer_map (outbuf, GST_MAP_READ, &rtp);
|
|
rtp_time = gst_rtp_buffer_get_timestamp (&rtp);
|
|
gst_rtp_buffer_unmap (&rtp);
|
|
|
|
GST_LOG_OBJECT (jitterbuffer, "rtp %" G_GUINT32_FORMAT ", ext %"
|
|
G_GUINT64_FORMAT, rtp_time, priv->ext_timestamp);
|
|
|
|
if (rtp_time < priv->ext_timestamp) {
|
|
ext_time = priv->ext_timestamp;
|
|
} else {
|
|
ext_time = gst_rtp_buffer_ext_timestamp (&priv->ext_timestamp, rtp_time);
|
|
}
|
|
|
|
if (ext_time > priv->clock_base)
|
|
elapsed = ext_time - priv->clock_base;
|
|
else
|
|
elapsed = 0;
|
|
|
|
elapsed = gst_util_uint64_scale_int (elapsed, GST_SECOND, priv->clock_rate);
|
|
return elapsed;
|
|
}
|
|
|
|
/*
|
|
* 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;
|
|
GstRTPBuffer rtp = { NULL, };
|
|
|
|
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))) {
|
|
GstClockTime elapsed, delay, left;
|
|
|
|
if (priv->estimated_eos == -1)
|
|
goto do_wait;
|
|
|
|
outbuf = rtp_jitter_buffer_peek (priv->jbuf);
|
|
if (outbuf != NULL) {
|
|
elapsed = compute_elapsed (jitterbuffer, outbuf);
|
|
if (GST_BUFFER_DURATION_IS_VALID (outbuf))
|
|
elapsed += GST_BUFFER_DURATION (outbuf);
|
|
} else {
|
|
GST_INFO_OBJECT (jitterbuffer, "no buffer, using last_elapsed");
|
|
elapsed = priv->last_elapsed;
|
|
}
|
|
|
|
delay = rtp_jitter_buffer_get_delay (priv->jbuf);
|
|
|
|
if (priv->estimated_eos > elapsed)
|
|
left = priv->estimated_eos - elapsed;
|
|
else
|
|
left = 0;
|
|
|
|
GST_INFO_OBJECT (jitterbuffer, "buffering, elapsed %" GST_TIME_FORMAT
|
|
" estimated_eos %" GST_TIME_FORMAT " left %" GST_TIME_FORMAT
|
|
" delay %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (elapsed), GST_TIME_ARGS (priv->estimated_eos),
|
|
GST_TIME_ARGS (left), GST_TIME_ARGS (delay));
|
|
if (left > delay)
|
|
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_INFO_OBJECT (jitterbuffer, "scheduling timeout");
|
|
id = gst_clock_new_single_shot_id (clock, sync_time);
|
|
gst_clock_id_wait_async (id, (GstClockCallback) eos_reached,
|
|
jitterbuffer, NULL);
|
|
}
|
|
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 */
|
|
gst_rtp_buffer_map (outbuf, GST_MAP_READ, &rtp);
|
|
seqnum = gst_rtp_buffer_get_seq (&rtp);
|
|
gst_rtp_buffer_unmap (&rtp);
|
|
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;
|
|
GstClockTimeDiff clock_jitter;
|
|
guint32 lost_packets = 1;
|
|
gboolean lost_packets_late = FALSE;
|
|
|
|
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;
|
|
}
|
|
|
|
/* prepare for sync against clock */
|
|
sync_time = get_sync_time (jitterbuffer, out_time);
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "sync to timestamp %" GST_TIME_FORMAT
|
|
" with sync time %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (out_time), GST_TIME_ARGS (sync_time));
|
|
|
|
/* create an entry for the clock */
|
|
id = priv->clock_id = gst_clock_new_single_shot_id (clock, sync_time);
|
|
priv->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, &clock_jitter);
|
|
|
|
if (ret == GST_CLOCK_EARLY && gap > 0
|
|
&& clock_jitter > (priv->latency_ns + priv->peer_latency)) {
|
|
GstClockTimeDiff total_duration;
|
|
GstClockTime out_time_diff;
|
|
|
|
out_time_diff = apply_offset (jitterbuffer, timestamp) - out_time;
|
|
total_duration = MIN (out_time_diff, clock_jitter);
|
|
|
|
if (duration > 0)
|
|
lost_packets = total_duration / duration;
|
|
else
|
|
lost_packets = gap;
|
|
total_duration = lost_packets * duration;
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer,
|
|
"Current sync_time has expired a long time ago (+%" GST_TIME_FORMAT
|
|
") Cover up %d lost packets with duration %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (clock_jitter),
|
|
lost_packets, GST_TIME_ARGS (total_duration));
|
|
|
|
duration = total_duration;
|
|
lost_packets_late = TRUE;
|
|
}
|
|
|
|
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 some packets. Create an event for this. */
|
|
if (lost_packets > 1)
|
|
GST_DEBUG_OBJECT (jitterbuffer, "Packets #%d -> #%d lost", next_seqnum,
|
|
next_seqnum + lost_packets - 1);
|
|
else
|
|
GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d lost", next_seqnum);
|
|
|
|
priv->num_late += lost_packets;
|
|
discont = TRUE;
|
|
|
|
/* update our expected next packet */
|
|
priv->last_popped_seqnum = next_seqnum;
|
|
priv->last_out_time += duration;
|
|
priv->next_seqnum = (next_seqnum + lost_packets) & 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,
|
|
"late", G_TYPE_BOOLEAN, lost_packets_late, 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);
|
|
|
|
check_buffering_percent (jitterbuffer, &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_PTS (outbuf) = out_time;
|
|
GST_BUFFER_DTS (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 elapsed, estimated;
|
|
|
|
elapsed = compute_elapsed (jitterbuffer, outbuf);
|
|
|
|
if (elapsed > priv->last_elapsed || !priv->last_elapsed) {
|
|
guint64 left;
|
|
|
|
priv->last_elapsed = elapsed;
|
|
|
|
left = priv->npt_stop - priv->npt_start;
|
|
GST_LOG_OBJECT (jitterbuffer, "left %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (left));
|
|
|
|
if (elapsed > 0)
|
|
estimated = gst_util_uint64_scale (out_time, left, elapsed);
|
|
else {
|
|
/* if there is almost nothing left,
|
|
* we may never advance enough to end up in the above case */
|
|
if (left < GST_SECOND)
|
|
estimated = GST_SECOND;
|
|
else
|
|
estimated = -1;
|
|
}
|
|
|
|
GST_LOG_OBJECT (jitterbuffer, "elapsed %" GST_TIME_FORMAT ", estimated %"
|
|
GST_TIME_FORMAT, GST_TIME_ARGS (elapsed), GST_TIME_ARGS (estimated));
|
|
|
|
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_EOS;
|
|
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, GstObject * parent,
|
|
GstBuffer * buffer)
|
|
{
|
|
GstRtpJitterBuffer *jitterbuffer;
|
|
GstRtpJitterBufferPrivate *priv;
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
guint64 base_rtptime, base_time;
|
|
guint32 clock_rate;
|
|
guint64 last_rtptime;
|
|
guint32 ssrc;
|
|
GstRTCPPacket packet;
|
|
guint64 ext_rtptime, diff;
|
|
guint32 rtptime;
|
|
gboolean drop = FALSE;
|
|
GstRTCPBuffer rtcp = { NULL, };
|
|
guint64 clock_base;
|
|
|
|
jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
|
|
|
|
if (G_UNLIKELY (!gst_rtcp_buffer_validate (buffer)))
|
|
goto invalid_buffer;
|
|
|
|
priv = jitterbuffer->priv;
|
|
|
|
gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
|
|
|
|
if (!gst_rtcp_buffer_get_first_packet (&rtcp, &packet))
|
|
goto empty_buffer;
|
|
|
|
/* first packet must be SR or RR or else the validate would have failed */
|
|
switch (gst_rtcp_packet_get_type (&packet)) {
|
|
case GST_RTCP_TYPE_SR:
|
|
gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, NULL, &rtptime,
|
|
NULL, NULL);
|
|
break;
|
|
default:
|
|
goto ignore_buffer;
|
|
}
|
|
gst_rtcp_buffer_unmap (&rtcp);
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "received RTCP of SSRC %08x", ssrc);
|
|
|
|
JBUF_LOCK (priv);
|
|
/* convert the RTP timestamp to our extended timestamp, using the same offset
|
|
* we used in the jitterbuffer */
|
|
ext_rtptime = priv->jbuf->ext_rtptime;
|
|
ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
|
|
|
|
/* get the last values from the jitterbuffer */
|
|
rtp_jitter_buffer_get_sync (priv->jbuf, &base_rtptime, &base_time,
|
|
&clock_rate, &last_rtptime);
|
|
|
|
clock_base = priv->clock_base;
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "ext SR %" G_GUINT64_FORMAT ", base %"
|
|
G_GUINT64_FORMAT ", clock-rate %" G_GUINT32_FORMAT
|
|
", clock-base %" G_GUINT64_FORMAT,
|
|
ext_rtptime, base_rtptime, clock_rate, clock_base);
|
|
|
|
if (base_rtptime == -1 || clock_rate == -1 || base_time == -1) {
|
|
GST_DEBUG_OBJECT (jitterbuffer, "dropping, no RTP values");
|
|
drop = TRUE;
|
|
} else {
|
|
/* we can't accept anything that happened before we did the last resync */
|
|
if (base_rtptime > ext_rtptime) {
|
|
GST_DEBUG_OBJECT (jitterbuffer, "dropping, older than base time");
|
|
drop = TRUE;
|
|
} else {
|
|
/* the SR RTP timestamp must be something close to what we last observed
|
|
* in the jitterbuffer */
|
|
if (ext_rtptime > last_rtptime) {
|
|
/* check how far ahead it is to our RTP timestamps */
|
|
diff = ext_rtptime - last_rtptime;
|
|
/* if bigger than 1 second, we drop it */
|
|
if (diff > clock_rate) {
|
|
GST_DEBUG_OBJECT (jitterbuffer, "too far ahead");
|
|
/* should drop this, but some RTSP servers end up with bogus
|
|
* way too ahead RTCP packet when repeated PAUSE/PLAY,
|
|
* so still trigger rptbin sync but invalidate RTCP data
|
|
* (sync might use other methods) */
|
|
ext_rtptime = -1;
|
|
}
|
|
GST_DEBUG_OBJECT (jitterbuffer, "ext last %" G_GUINT64_FORMAT ", diff %"
|
|
G_GUINT64_FORMAT, last_rtptime, diff);
|
|
}
|
|
}
|
|
}
|
|
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, base_time,
|
|
"clock-rate", G_TYPE_UINT, clock_rate,
|
|
"clock-base", G_TYPE_UINT64, clock_base,
|
|
"sr-ext-rtptime", G_TYPE_UINT64, ext_rtptime,
|
|
"sr-buffer", GST_TYPE_BUFFER, 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);
|
|
|
|
return ret;
|
|
|
|
invalid_buffer:
|
|
{
|
|
/* this is not fatal but should be filtered earlier */
|
|
GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
|
|
("Received invalid RTCP payload, dropping"));
|
|
ret = GST_FLOW_OK;
|
|
goto done;
|
|
}
|
|
empty_buffer:
|
|
{
|
|
/* this is not fatal but should be filtered earlier */
|
|
GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
|
|
("Received empty RTCP payload, dropping"));
|
|
gst_rtcp_buffer_unmap (&rtcp);
|
|
ret = GST_FLOW_OK;
|
|
goto done;
|
|
}
|
|
ignore_buffer:
|
|
{
|
|
GST_DEBUG_OBJECT (jitterbuffer, "ignoring RTCP packet");
|
|
gst_rtcp_buffer_unmap (&rtcp);
|
|
ret = GST_FLOW_OK;
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_jitter_buffer_sink_query (GstPad * pad, GstObject * parent,
|
|
GstQuery * query)
|
|
{
|
|
gboolean res = FALSE;
|
|
|
|
switch (GST_QUERY_TYPE (query)) {
|
|
case GST_QUERY_CAPS:
|
|
{
|
|
GstCaps *filter, *caps;
|
|
|
|
gst_query_parse_caps (query, &filter);
|
|
caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
|
|
gst_query_set_caps_result (query, caps);
|
|
gst_caps_unref (caps);
|
|
res = TRUE;
|
|
break;
|
|
}
|
|
default:
|
|
if (GST_QUERY_IS_SERIALIZED (query)) {
|
|
GST_WARNING_OBJECT (pad, "unhandled serialized query");
|
|
res = FALSE;
|
|
} else {
|
|
res = gst_pad_query_default (pad, parent, query);
|
|
}
|
|
break;
|
|
}
|
|
return res;
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_jitter_buffer_src_query (GstPad * pad, GstObject * parent,
|
|
GstQuery * query)
|
|
{
|
|
GstRtpJitterBuffer *jitterbuffer;
|
|
GstRtpJitterBufferPrivate *priv;
|
|
gboolean res = FALSE;
|
|
|
|
jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
|
|
priv = jitterbuffer->priv;
|
|
|
|
switch (GST_QUERY_TYPE (query)) {
|
|
case GST_QUERY_LATENCY:
|
|
{
|
|
/* We need to send the query upstream and add the returned latency to our
|
|
* own */
|
|
GstClockTime min_latency, max_latency;
|
|
gboolean us_live;
|
|
GstClockTime our_latency;
|
|
|
|
if ((res = gst_pad_peer_query (priv->sinkpad, query))) {
|
|
gst_query_parse_latency (query, &us_live, &min_latency, &max_latency);
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "Peer latency: min %"
|
|
GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
|
|
|
|
/* store this so that we can safely sync on the peer buffers. */
|
|
JBUF_LOCK (priv);
|
|
priv->peer_latency = min_latency;
|
|
our_latency = priv->latency_ns;
|
|
JBUF_UNLOCK (priv);
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "Our latency: %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (our_latency));
|
|
|
|
/* we add some latency but can buffer an infinite amount of time */
|
|
min_latency += our_latency;
|
|
max_latency = -1;
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "Calculated total latency : min %"
|
|
GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
|
|
|
|
gst_query_set_latency (query, TRUE, min_latency, max_latency);
|
|
}
|
|
break;
|
|
}
|
|
case GST_QUERY_POSITION:
|
|
{
|
|
GstClockTime start, last_out;
|
|
GstFormat fmt;
|
|
|
|
gst_query_parse_position (query, &fmt, NULL);
|
|
if (fmt != GST_FORMAT_TIME) {
|
|
res = gst_pad_query_default (pad, parent, query);
|
|
break;
|
|
}
|
|
|
|
JBUF_LOCK (priv);
|
|
start = priv->npt_start;
|
|
last_out = priv->last_out_time;
|
|
JBUF_UNLOCK (priv);
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "npt start %" GST_TIME_FORMAT
|
|
", last out %" GST_TIME_FORMAT, GST_TIME_ARGS (start),
|
|
GST_TIME_ARGS (last_out));
|
|
|
|
if (GST_CLOCK_TIME_IS_VALID (start) && GST_CLOCK_TIME_IS_VALID (last_out)) {
|
|
/* bring 0-based outgoing time to stream time */
|
|
gst_query_set_position (query, GST_FORMAT_TIME, start + last_out);
|
|
res = TRUE;
|
|
} else {
|
|
res = gst_pad_query_default (pad, parent, query);
|
|
}
|
|
break;
|
|
}
|
|
case GST_QUERY_CAPS:
|
|
{
|
|
GstCaps *filter, *caps;
|
|
|
|
gst_query_parse_caps (query, &filter);
|
|
caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
|
|
gst_query_set_caps_result (query, caps);
|
|
gst_caps_unref (caps);
|
|
res = TRUE;
|
|
break;
|
|
}
|
|
default:
|
|
res = gst_pad_query_default (pad, parent, query);
|
|
break;
|
|
}
|
|
|
|
return res;
|
|
}
|
|
|
|
static void
|
|
gst_rtp_jitter_buffer_set_property (GObject * object,
|
|
guint prop_id, const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstRtpJitterBuffer *jitterbuffer;
|
|
GstRtpJitterBufferPrivate *priv;
|
|
|
|
jitterbuffer = GST_RTP_JITTER_BUFFER (object);
|
|
priv = jitterbuffer->priv;
|
|
|
|
switch (prop_id) {
|
|
case PROP_LATENCY:
|
|
{
|
|
guint new_latency, old_latency;
|
|
|
|
new_latency = g_value_get_uint (value);
|
|
|
|
JBUF_LOCK (priv);
|
|
old_latency = priv->latency_ms;
|
|
priv->latency_ms = new_latency;
|
|
priv->latency_ns = priv->latency_ms * GST_MSECOND;
|
|
rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
|
|
JBUF_UNLOCK (priv);
|
|
|
|
/* post message if latency changed, this will inform the parent pipeline
|
|
* that a latency reconfiguration is possible/needed. */
|
|
if (new_latency != old_latency) {
|
|
GST_DEBUG_OBJECT (jitterbuffer, "latency changed to: %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (new_latency * GST_MSECOND));
|
|
|
|
gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer),
|
|
gst_message_new_latency (GST_OBJECT_CAST (jitterbuffer)));
|
|
}
|
|
break;
|
|
}
|
|
case PROP_DROP_ON_LATENCY:
|
|
JBUF_LOCK (priv);
|
|
priv->drop_on_latency = g_value_get_boolean (value);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_TS_OFFSET:
|
|
JBUF_LOCK (priv);
|
|
priv->ts_offset = g_value_get_int64 (value);
|
|
/* 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;
|
|
}
|
|
}
|