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d704790519
Element sections were not rendered anymore after the hotdoc port, fixing this revealed a few incorrect links.
526 lines
15 KiB
C
526 lines
15 KiB
C
/* RTP Retransmission queue element for GStreamer
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*
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* gstrtprtxqueue.c:
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*
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* Copyright (C) 2013 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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* SECTION:element-rtprtxqueue
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* @title: rtprtxqueue
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*
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* rtprtxqueue maintains a queue of transmitted RTP packets, up to a
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* configurable limit (see #GstRTPRtxQueue:max-size-time,
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* #GstRTPRtxQueue:max-size-packets), and retransmits them upon request
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* from the downstream rtpsession (GstRTPRetransmissionRequest event).
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*
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* This element is similar to rtprtxsend, but it has differences:
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* - Retransmission from rtprtxqueue is not RFC 4588 compliant. The
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* retransmitted packets have the same ssrc and payload type as the original
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* stream.
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* - As a side-effect of the above, rtprtxqueue does not require the use of
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* rtprtxreceive on the receiving end. rtpjitterbuffer alone is able to
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* reconstruct the stream.
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* - Retransmission from rtprtxqueue happens as soon as the next regular flow
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* packet is chained, while rtprtxsend retransmits as soon as the retransmission
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* event is received, using a helper thread.
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* - rtprtxqueue can be used with rtpbin without the need of hooking to its
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* #GstRtpBin::request-aux-sender signal, which means it can be used with
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* rtpbin using gst-launch.
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*
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* See also #GstRtpRtxSend, #GstRtpRtxReceive
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*
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* # Example pipelines
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*
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* |[
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* gst-launch-1.0 rtpbin name=b rtp-profile=avpf \
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* audiotestsrc is-live=true ! opusenc ! rtpopuspay pt=96 ! rtprtxqueue ! b.send_rtp_sink_0 \
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* b.send_rtp_src_0 ! identity drop-probability=0.01 ! udpsink host="127.0.0.1" port=5000 \
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* udpsrc port=5001 ! b.recv_rtcp_sink_0 \
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* b.send_rtcp_src_0 ! udpsink host="127.0.0.1" port=5002 sync=false async=false
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* ]|
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* Sender pipeline
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*
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* |[
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* gst-launch-1.0 rtpbin name=b rtp-profile=avpf do-retransmission=true \
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* udpsrc port=5000 caps="application/x-rtp,media=(string)audio,clock-rate=(int)48000,encoding-name=(string)OPUS,payload=(int)96" ! \
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* b.recv_rtp_sink_0 \
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* b. ! rtpopusdepay ! opusdec ! audioconvert ! audioresample ! autoaudiosink \
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* udpsrc port=5002 ! b.recv_rtcp_sink_0 \
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* b.send_rtcp_src_0 ! udpsink host="127.0.0.1" port=5001 sync=false async=false
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* ]|
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* Receiver pipeline
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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 <gst/gst.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include <string.h>
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#include "gstrtprtxqueue.h"
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GST_DEBUG_CATEGORY_STATIC (gst_rtp_rtx_queue_debug);
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#define GST_CAT_DEFAULT gst_rtp_rtx_queue_debug
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#define DEFAULT_MAX_SIZE_TIME 0
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#define DEFAULT_MAX_SIZE_PACKETS 100
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enum
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{
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PROP_0,
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PROP_MAX_SIZE_TIME,
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PROP_MAX_SIZE_PACKETS,
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PROP_REQUESTS,
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PROP_FULFILLED_REQUESTS,
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};
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static GstStaticPadTemplate src_factory = 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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);
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static GstStaticPadTemplate sink_factory = 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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);
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static gboolean gst_rtp_rtx_queue_src_event (GstPad * pad, GstObject * parent,
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GstEvent * event);
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static gboolean gst_rtp_rtx_queue_sink_event (GstPad * pad, GstObject * parent,
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GstEvent * event);
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static GstFlowReturn gst_rtp_rtx_queue_chain (GstPad * pad, GstObject * parent,
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GstBuffer * buffer);
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static GstFlowReturn gst_rtp_rtx_queue_chain_list (GstPad * pad,
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GstObject * parent, GstBufferList * list);
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static GstStateChangeReturn gst_rtp_rtx_queue_change_state (GstElement *
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element, GstStateChange transition);
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static void gst_rtp_rtx_queue_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_rtp_rtx_queue_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static void gst_rtp_rtx_queue_finalize (GObject * object);
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G_DEFINE_TYPE (GstRTPRtxQueue, gst_rtp_rtx_queue, GST_TYPE_ELEMENT);
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static void
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gst_rtp_rtx_queue_class_init (GstRTPRtxQueueClass * 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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gobject_class->get_property = gst_rtp_rtx_queue_get_property;
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gobject_class->set_property = gst_rtp_rtx_queue_set_property;
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gobject_class->finalize = gst_rtp_rtx_queue_finalize;
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g_object_class_install_property (gobject_class, PROP_MAX_SIZE_TIME,
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g_param_spec_uint ("max-size-time", "Max Size Times",
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"Amount of ms to queue (0 = unlimited)", 0, G_MAXUINT,
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DEFAULT_MAX_SIZE_TIME, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_MAX_SIZE_PACKETS,
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g_param_spec_uint ("max-size-packets", "Max Size Packets",
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"Amount of packets to queue (0 = unlimited)", 0, G_MAXUINT,
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DEFAULT_MAX_SIZE_PACKETS,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_REQUESTS,
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g_param_spec_uint ("requests", "Requests",
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"Total number of retransmission requests", 0, G_MAXUINT,
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0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_FULFILLED_REQUESTS,
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g_param_spec_uint ("fulfilled-requests", "Fulfilled Requests",
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"Number of fulfilled retransmission requests", 0, G_MAXUINT,
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0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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gst_element_class_add_static_pad_template (gstelement_class, &src_factory);
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gst_element_class_add_static_pad_template (gstelement_class, &sink_factory);
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gst_element_class_set_static_metadata (gstelement_class,
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"RTP Retransmission Queue", "Codec",
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"Keep RTP packets in a queue for retransmission",
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"Wim Taymans <wim.taymans@gmail.com>");
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gstelement_class->change_state =
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GST_DEBUG_FUNCPTR (gst_rtp_rtx_queue_change_state);
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}
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static void
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gst_rtp_rtx_queue_reset (GstRTPRtxQueue * rtx, gboolean full)
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{
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g_mutex_lock (&rtx->lock);
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g_queue_foreach (rtx->queue, (GFunc) gst_buffer_unref, NULL);
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g_queue_clear (rtx->queue);
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g_list_foreach (rtx->pending, (GFunc) gst_buffer_unref, NULL);
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g_list_free (rtx->pending);
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rtx->pending = NULL;
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rtx->n_requests = 0;
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rtx->n_fulfilled_requests = 0;
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g_mutex_unlock (&rtx->lock);
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}
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static void
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gst_rtp_rtx_queue_finalize (GObject * object)
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{
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GstRTPRtxQueue *rtx = GST_RTP_RTX_QUEUE (object);
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gst_rtp_rtx_queue_reset (rtx, TRUE);
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g_queue_free (rtx->queue);
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g_mutex_clear (&rtx->lock);
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G_OBJECT_CLASS (gst_rtp_rtx_queue_parent_class)->finalize (object);
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}
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static void
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gst_rtp_rtx_queue_init (GstRTPRtxQueue * rtx)
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{
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GstElementClass *klass = GST_ELEMENT_GET_CLASS (rtx);
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rtx->srcpad =
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gst_pad_new_from_template (gst_element_class_get_pad_template (klass,
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"src"), "src");
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GST_PAD_SET_PROXY_CAPS (rtx->srcpad);
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GST_PAD_SET_PROXY_ALLOCATION (rtx->srcpad);
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gst_pad_set_event_function (rtx->srcpad,
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GST_DEBUG_FUNCPTR (gst_rtp_rtx_queue_src_event));
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gst_element_add_pad (GST_ELEMENT (rtx), rtx->srcpad);
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rtx->sinkpad =
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gst_pad_new_from_template (gst_element_class_get_pad_template (klass,
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"sink"), "sink");
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GST_PAD_SET_PROXY_CAPS (rtx->sinkpad);
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GST_PAD_SET_PROXY_ALLOCATION (rtx->sinkpad);
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gst_pad_set_event_function (rtx->sinkpad,
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GST_DEBUG_FUNCPTR (gst_rtp_rtx_queue_sink_event));
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gst_pad_set_chain_function (rtx->sinkpad,
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GST_DEBUG_FUNCPTR (gst_rtp_rtx_queue_chain));
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gst_pad_set_chain_list_function (rtx->sinkpad,
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GST_DEBUG_FUNCPTR (gst_rtp_rtx_queue_chain_list));
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gst_element_add_pad (GST_ELEMENT (rtx), rtx->sinkpad);
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rtx->queue = g_queue_new ();
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g_mutex_init (&rtx->lock);
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rtx->max_size_time = DEFAULT_MAX_SIZE_TIME;
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rtx->max_size_packets = DEFAULT_MAX_SIZE_PACKETS;
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}
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typedef struct
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{
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GstRTPRtxQueue *rtx;
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guint seqnum;
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gboolean found;
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} RTXData;
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static void
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push_seqnum (GstBuffer * buffer, RTXData * data)
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{
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GstRTPRtxQueue *rtx = data->rtx;
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GstRTPBuffer rtpbuffer = GST_RTP_BUFFER_INIT;
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guint16 seqnum;
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if (data->found)
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return;
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if (!GST_IS_BUFFER (buffer) ||
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!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtpbuffer))
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return;
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seqnum = gst_rtp_buffer_get_seq (&rtpbuffer);
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gst_rtp_buffer_unmap (&rtpbuffer);
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if (seqnum == data->seqnum) {
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data->found = TRUE;
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GST_DEBUG_OBJECT (rtx, "found %d", seqnum);
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rtx->pending = g_list_prepend (rtx->pending, gst_buffer_ref (buffer));
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}
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}
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static gboolean
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gst_rtp_rtx_queue_src_event (GstPad * pad, GstObject * parent, GstEvent * event)
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{
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GstRTPRtxQueue *rtx = GST_RTP_RTX_QUEUE (parent);
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gboolean res;
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_CUSTOM_UPSTREAM:
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{
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const GstStructure *s;
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s = gst_event_get_structure (event);
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if (gst_structure_has_name (s, "GstRTPRetransmissionRequest")) {
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guint seqnum;
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RTXData data;
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if (!gst_structure_get_uint (s, "seqnum", &seqnum))
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seqnum = -1;
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GST_DEBUG_OBJECT (rtx, "request %d", seqnum);
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g_mutex_lock (&rtx->lock);
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data.rtx = rtx;
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data.seqnum = seqnum;
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data.found = FALSE;
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rtx->n_requests += 1;
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g_queue_foreach (rtx->queue, (GFunc) push_seqnum, &data);
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g_mutex_unlock (&rtx->lock);
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gst_event_unref (event);
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res = TRUE;
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} else {
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res = gst_pad_event_default (pad, parent, event);
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}
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break;
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}
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default:
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res = gst_pad_event_default (pad, parent, event);
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break;
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}
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return res;
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}
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static gboolean
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gst_rtp_rtx_queue_sink_event (GstPad * pad, GstObject * parent,
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GstEvent * event)
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{
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GstRTPRtxQueue *rtx = GST_RTP_RTX_QUEUE (parent);
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gboolean res;
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_SEGMENT:
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{
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g_mutex_lock (&rtx->lock);
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gst_event_copy_segment (event, &rtx->head_segment);
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g_queue_push_head (rtx->queue, gst_event_ref (event));
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g_mutex_unlock (&rtx->lock);
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/* fall through */
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}
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default:
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res = gst_pad_event_default (pad, parent, event);
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break;
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}
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return res;
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}
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static void
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do_push (GstBuffer * buffer, GstRTPRtxQueue * rtx)
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{
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rtx->n_fulfilled_requests += 1;
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gst_pad_push (rtx->srcpad, buffer);
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}
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static guint32
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get_ts_diff (GstRTPRtxQueue * rtx)
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{
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GstClockTime high_ts, low_ts;
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GstClockTimeDiff result;
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GstBuffer *high_buf, *low_buf;
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high_buf = g_queue_peek_head (rtx->queue);
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while (GST_IS_EVENT ((low_buf = g_queue_peek_tail (rtx->queue)))) {
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GstEvent *event = g_queue_pop_tail (rtx->queue);
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gst_event_copy_segment (event, &rtx->tail_segment);
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gst_event_unref (event);
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}
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if (!high_buf || !low_buf || high_buf == low_buf)
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return 0;
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high_ts = GST_BUFFER_TIMESTAMP (high_buf);
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low_ts = GST_BUFFER_TIMESTAMP (low_buf);
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high_ts = gst_segment_to_running_time (&rtx->head_segment, GST_FORMAT_TIME,
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high_ts);
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low_ts = gst_segment_to_running_time (&rtx->tail_segment, GST_FORMAT_TIME,
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low_ts);
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result = high_ts - low_ts;
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/* return value in ms instead of ns */
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return (guint32) gst_util_uint64_scale_int (result, 1, GST_MSECOND);
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}
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/* Must be called with rtx->lock */
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static void
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shrink_queue (GstRTPRtxQueue * rtx)
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{
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if (rtx->max_size_packets) {
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while (g_queue_get_length (rtx->queue) > rtx->max_size_packets)
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gst_buffer_unref (g_queue_pop_tail (rtx->queue));
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}
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if (rtx->max_size_time) {
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while (get_ts_diff (rtx) > rtx->max_size_time)
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gst_buffer_unref (g_queue_pop_tail (rtx->queue));
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}
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}
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static GstFlowReturn
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gst_rtp_rtx_queue_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
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{
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GstRTPRtxQueue *rtx;
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GstFlowReturn ret;
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GList *pending;
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rtx = GST_RTP_RTX_QUEUE (parent);
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g_mutex_lock (&rtx->lock);
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g_queue_push_head (rtx->queue, gst_buffer_ref (buffer));
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shrink_queue (rtx);
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pending = rtx->pending;
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rtx->pending = NULL;
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g_mutex_unlock (&rtx->lock);
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pending = g_list_reverse (pending);
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g_list_foreach (pending, (GFunc) do_push, rtx);
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g_list_free (pending);
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ret = gst_pad_push (rtx->srcpad, buffer);
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return ret;
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}
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static gboolean
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push_to_queue (GstBuffer ** buffer, guint idx, gpointer user_data)
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{
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GQueue *queue = user_data;
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g_queue_push_head (queue, gst_buffer_ref (*buffer));
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return TRUE;
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}
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static GstFlowReturn
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gst_rtp_rtx_queue_chain_list (GstPad * pad, GstObject * parent,
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GstBufferList * list)
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{
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GstRTPRtxQueue *rtx;
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GstFlowReturn ret;
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GList *pending;
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rtx = GST_RTP_RTX_QUEUE (parent);
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g_mutex_lock (&rtx->lock);
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gst_buffer_list_foreach (list, push_to_queue, rtx->queue);
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shrink_queue (rtx);
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pending = rtx->pending;
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rtx->pending = NULL;
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g_mutex_unlock (&rtx->lock);
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pending = g_list_reverse (pending);
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g_list_foreach (pending, (GFunc) do_push, rtx);
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g_list_free (pending);
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ret = gst_pad_push_list (rtx->srcpad, list);
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return ret;
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}
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static void
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gst_rtp_rtx_queue_get_property (GObject * object,
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guint prop_id, GValue * value, GParamSpec * pspec)
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{
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GstRTPRtxQueue *rtx = GST_RTP_RTX_QUEUE (object);
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switch (prop_id) {
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case PROP_MAX_SIZE_TIME:
|
|
g_value_set_uint (value, rtx->max_size_time);
|
|
break;
|
|
case PROP_MAX_SIZE_PACKETS:
|
|
g_value_set_uint (value, rtx->max_size_packets);
|
|
break;
|
|
case PROP_REQUESTS:
|
|
g_value_set_uint (value, rtx->n_requests);
|
|
break;
|
|
case PROP_FULFILLED_REQUESTS:
|
|
g_value_set_uint (value, rtx->n_fulfilled_requests);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtp_rtx_queue_set_property (GObject * object,
|
|
guint prop_id, const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstRTPRtxQueue *rtx = GST_RTP_RTX_QUEUE (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_MAX_SIZE_TIME:
|
|
rtx->max_size_time = g_value_get_uint (value);
|
|
break;
|
|
case PROP_MAX_SIZE_PACKETS:
|
|
rtx->max_size_packets = g_value_get_uint (value);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_rtp_rtx_queue_change_state (GstElement * element, GstStateChange transition)
|
|
{
|
|
GstStateChangeReturn ret;
|
|
GstRTPRtxQueue *rtx;
|
|
|
|
rtx = GST_RTP_RTX_QUEUE (element);
|
|
|
|
switch (transition) {
|
|
default:
|
|
break;
|
|
}
|
|
|
|
ret =
|
|
GST_ELEMENT_CLASS (gst_rtp_rtx_queue_parent_class)->change_state (element,
|
|
transition);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
gst_rtp_rtx_queue_reset (rtx, TRUE);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
gboolean
|
|
gst_rtp_rtx_queue_plugin_init (GstPlugin * plugin)
|
|
{
|
|
GST_DEBUG_CATEGORY_INIT (gst_rtp_rtx_queue_debug, "rtprtxqueue", 0,
|
|
"rtp retransmission queue");
|
|
|
|
return gst_element_register (plugin, "rtprtxqueue", GST_RANK_NONE,
|
|
GST_TYPE_RTP_RTX_QUEUE);
|
|
}
|