Commit 5a1aa759 authored by Julien Isorce's avatar Julien Isorce Committed by Wim Taymans

rtpmanager: add new rtprtxsend / rtprtxreceive elements

The purpose of the sender RTX object is to keep a history
of RTP packets up to a configurable limit (in time). It will
listen for custom retransmission events from downstream. When
it receives a request for retransmission, it will look up the
requested seqnum in its list of stored packets. If the packet
is available, it will create a RTX packet according to RFC 4588
and send this as an auxiliary stream.

The receiver will listen to the custom retransmission events
from the downstream jitterbuffer and will remember the SSRC1
of the stream and seqnum that was requested. When it sees a
packet with one of the stored seqnum, it associates the SSRC2
of the stream with the SSRC1 of the master stream. From then
on it knows that SSRC2 is the retransmission stream of SSRC1.
This algorithm is stated in RFC 4588. For this algorithm to
work, RFC4588 also states that no two pending retransmission
requests can exist for the same seqnum and different SSRCs or
else it would be impossible to associate the retransmission with
the original requester SSRC.
When the RTX receiver has associated the retransmission packets,
it can depayload and forward them to the source pad of the element.

RTX is SSRC-multiplexed

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711084
parent 19c0e920
......@@ -10,6 +10,8 @@ libgstrtpmanager_la_SOURCES = gstrtpmanager.c \
gstrtpmux.c \
gstrtpptdemux.c \
gstrtprtxqueue.c \
gstrtprtxreceive.c \
gstrtprtxsend.c \
gstrtpssrcdemux.c \
rtpjitterbuffer.c \
rtpsession.c \
......@@ -24,6 +26,8 @@ noinst_HEADERS = gstrtpbin.h \
gstrtpptdemux.h \
gstrtpssrcdemux.h \
gstrtprtxqueue.h \
gstrtprtxreceive.h \
gstrtprtxsend.h \
rtpjitterbuffer.h \
rtpsession.h \
rtpsource.h \
......
......@@ -26,6 +26,8 @@
#include "gstrtpptdemux.h"
#include "gstrtpsession.h"
#include "gstrtprtxqueue.h"
#include "gstrtprtxreceive.h"
#include "gstrtprtxsend.h"
#include "gstrtpssrcdemux.h"
#include "gstrtpdtmfmux.h"
#include "gstrtpmux.h"
......@@ -51,6 +53,12 @@ plugin_init (GstPlugin * plugin)
if (!gst_rtp_rtx_queue_plugin_init (plugin))
return FALSE;
if (!gst_rtp_rtx_receive_plugin_init (plugin))
return FALSE;
if (!gst_rtp_rtx_send_plugin_init (plugin))
return FALSE;
if (!gst_element_register (plugin, "rtpssrcdemux", GST_RANK_NONE,
GST_TYPE_RTP_SSRC_DEMUX))
return FALSE;
......
/* RTP Retransmission receiver element for GStreamer
*
* gstrtprtxreceive.c:
*
* Copyright (C) 2013 Collabora Ltd.
* @author Julien Isorce <julien.isorce@collabora.co.uk>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-rtprtxreceive
* @see_also: rtprtxsend, rtpsession, rtpjitterbuffer
*
* The receiver will listen to the custom retransmission events from the
* downstream jitterbuffer and will remember the SSRC1 of the stream and
* seqnum that was requested. When it sees a packet with one of the stored
* seqnum, it associates the SSRC2 of the stream with the SSRC1 of the
* master stream. From then it knows that SSRC2 is the retransmission
* stream of SSRC1. This algorithm is stated in RFC 4588. For this
* algorithm to work, RFC4588 also states that no two pending retransmission
* requests can exist for the same seqnum and different SSRCs or else it
* would be impossible to associate the retransmission with the original
* requester SSRC.
* When the RTX receiver has associated the retransmission packets,
* it can depayload and forward them to the source pad of the element.
* RTX is SSRC-multiplexed. See #GstRtpRtxSend
*
* <refsect2>
* <title>Example pipelines</title>
* |[
* gst-launch-1.0 rtpsession name=rtpsession \
* audiotestsrc ! speexenc ! rtpspeexpay pt=97 ! rtprtxsend rtx-payload-type=99 ! \
* identity drop-probability=0.1 ! rtpsession.send_rtp_sink \
* rtpsession.send_rtp_src ! udpsink host="127.0.0.1" port=5000 \
* udpsrc port=5001 ! rtpsession.recv_rtcp_sink \
* rtpsession.send_rtcp_src ! udpsink host="127.0.0.1" port=5002 sync=false async=false
* ]| Send audio stream through port 5000. (5001 and 5002 are just the rtcp link with the receiver)
* |[
* gst-launch-1.0 rtpsession name=rtpsession \
* udpsrc port=5000 caps="application/x-rtp,media=(string)audio,clock-rate=(int)44100,encoding-name=(string)SPEEX,encoding-params=(string)1,octet-align=(string)1" ! \
* rtpsession.recv_rtp_sink \
* rtpsession.recv_rtp_src ! rtprtxreceive rtx-payload-types="99" ! rtpjitterbuffer do-retransmission=true ! rtpspeexdepay ! \
* speexdec ! audioconvert ! autoaudiosink \
* rtpsession.send_rtcp_src ! udpsink host="127.0.0.1" port=5001 \
* udpsrc port=5002 ! rtpsession.recv_rtcp_sink sync=fakse async=false
* ]| Receive audio stream from port 5000. (5001 and 5002 are just the rtcp link with the sender)
* On sender side make sure to use a different payload type for the stream and
* its associated retransmission stream (see #GstRtpRtxSend). Note that several retransmission streams can
* have the same payload type so this is not deterministic. Actually the
* rtprtxreceiver element does the association using seqnum values.
* On receiver side set all the retransmission payload types (Those informations are retrieve
* through SDP).
* You should still hear a clear sound when setting drop-probability to something greater than 0.
* The rtpjitterbuffer will generate a custom upstream event GstRTPRetransmissionRequest when
* it assumes that one packet is missing. Then this request is translated to a FB NACK in the rtcp link
* Finally the rtpsession of the sender side re-convert it in a GstRTPRetransmissionRequest that will
* be handle by rtprtxsend.
* When increasing this value it may be possible that even the retransmission stream would be dropped
* so the receiver will ask to resend the packets again and again until it actually receive them.
* If the value is too high the rtprtxsend will not be able to retrieve the packet in its list of
* stored packets. For learning purpose you could try to increase the max-size-packets or max-size-time
* rtprtxsender's properties.
* Also note that you should use rtprtxsend through rtpbin and its set-aux-send property. See #GstRtpBin.
* |[
* gst-launch-1.0 rtpsession name=rtpsession0 \
* audiotestsrc wave=0 ! speexenc ! rtpspeexpay pt=97 ! rtprtxsend rtx-payload-type=99 seqnum-offset=1 ! \
* identity drop-probability=0.1 ! rtpsession0.send_rtp_sink \
* rtpsession0.send_rtp_src ! udpsink host="127.0.0.1" port=5000 \
* udpsrc port=5001 ! rtpsession0.recv_rtcp_sink \
* rtpsession0.send_rtcp_src ! udpsink host="127.0.0.1" port=5002 sync=false async=false \
* rtpsession name=rtpsession1 \
* audiotestsrc wave=0 ! speexenc ! rtpspeexpay pt=97 ! rtprtxsend rtx-payload-type=99 seqnum-offset=10 ! \
* identity drop-probability=0.1 ! rtpsession1.send_rtp_sink \
* rtpsession1.send_rtp_src ! udpsink host="127.0.0.1" port=5000 \
* udpsrc port=5004 ! rtpsession1.recv_rtcp_sink \
* rtpsession1.send_rtcp_src ! udpsink host="127.0.0.1" port=5002 sync=false async=false
* ]| Send two audio streams to port 5000.
* |[
* gst-launch-1.0 rtpsession name=rtpsession
* udpsrc port=5000 caps="application/x-rtp,media=(string)audio,clock-rate=(int)44100,encoding-name=(string)SPEEX,encoding-params=(string)1,octet-align=(string)1" ! \
* rtpsession.recv_rtp_sink \
* rtpsession.recv_rtp_src ! rtprtxreceive rtx-payload-types="99" ! rtpssrcdemux name=demux \
* demux. ! queue ! rtpjitterbuffer do-retransmission=true ! rtpspeexdepay ! speexdec ! audioconvert ! autoaudiosink \
* demux. ! queue ! rtpjitterbuffer do-retransmission=true ! rtpspeexdepay ! speexdec ! audioconvert ! autoaudiosink \
* rtpsession.send_rtcp_src ! ! tee name=t ! queue ! udpsink host="127.0.0.1" port=5001 t. ! queue ! udpsink host="127.0.0.1" port=5004 \
* udpsrc port=5002 ! rtpsession.recv_rtcp_sink sync=fakse async=false
* ]| Receive audio stream from port 5000.
* On sender side the two streams have the same payload type for master streams, Same about retransmission streams.
* The streams are sent to the network through two distincts sessions.
* But we need to set a different seqnum-offset to make sure their seqnum navigate at a different rate like in concrete cases.
* We could also choose the same seqnum offset but we would require to set a different initial seqnum value.
* This is also why the rtprtxreceive can succeed to do the association between master and retransmission stream.
* On receiver side the same session is used to receive the two streams. So the rtpssrcdemux is here to demultiplex
* those two streams. The rtprtxreceive is responsible for reconstructing the original packets from the two retransmission streams.
* You can play with the drop-probability value for one or both streams.
* You should hear a clear sound. (after a few seconds the two streams wave feel synchronized)
* </refsect2>
*
* Last reviewed on 2013-11-08 (1.x)
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst.h>
#include <gst/rtp/gstrtpbuffer.h>
#include <string.h>
#include "gstrtprtxreceive.h"
GST_DEBUG_CATEGORY_STATIC (gst_rtp_rtx_receive_debug);
#define GST_CAT_DEFAULT gst_rtp_rtx_receive_debug
#define DEFAULT_RTX_PAYLOAD_TYPES ""
enum
{
PROP_0,
PROP_RTX_PAYLOAD_TYPES,
PROP_NUM_RTX_REQUESTS,
PROP_NUM_RTX_PACKETS,
PROP_NUM_RTX_ASSOC_PACKETS,
PROP_LAST
};
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp")
);
static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp")
);
static gboolean gst_rtp_rtx_receive_src_event (GstPad * pad, GstObject * parent,
GstEvent * event);
static GstFlowReturn gst_rtp_rtx_receive_chain (GstPad * pad,
GstObject * parent, GstBuffer * buffer);
static GstStateChangeReturn gst_rtp_rtx_receive_change_state (GstElement *
element, GstStateChange transition);
static void gst_rtp_rtx_receive_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_rtp_rtx_receive_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void gst_rtp_rtx_receive_finalize (GObject * object);
G_DEFINE_TYPE (GstRtpRtxReceive, gst_rtp_rtx_receive, GST_TYPE_ELEMENT);
static void
gst_rtp_rtx_receive_class_init (GstRtpRtxReceiveClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gobject_class->get_property = gst_rtp_rtx_receive_get_property;
gobject_class->set_property = gst_rtp_rtx_receive_set_property;
gobject_class->finalize = gst_rtp_rtx_receive_finalize;
g_object_class_install_property (gobject_class, PROP_RTX_PAYLOAD_TYPES,
g_param_spec_string ("rtx-payload-types",
"Colon separated list of payload format type",
"Set through SDP (fmtp), it helps to detect restransmission streams "
"eg 97:101:127", DEFAULT_RTX_PAYLOAD_TYPES,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_NUM_RTX_REQUESTS,
g_param_spec_uint ("num-rtx-requests", "Num RTX Requests",
"Number of retransmission events received", 0, G_MAXUINT,
0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_NUM_RTX_PACKETS,
g_param_spec_uint ("num-rtx-packets", "Num RTX Packets",
" Number of retransmission packets received", 0, G_MAXUINT,
0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_NUM_RTX_ASSOC_PACKETS,
g_param_spec_uint ("num-rtx-assoc-packets",
"Num RTX Associated Packets", "Number of retransmission packets "
"correctly associated with retransmission requests", 0, G_MAXUINT,
0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&src_factory));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&sink_factory));
gst_element_class_set_static_metadata (gstelement_class,
"RTP Retransmission receiver", "Codec",
"Receive retransmitted RTP packets according to RFC4588",
"Julien Isorce <julien.isorce@collabora.co.uk>");
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_rtp_rtx_receive_change_state);
}
static void
gst_rtp_rtx_receive_reset (GstRtpRtxReceive * rtx)
{
g_mutex_lock (&rtx->lock);
g_hash_table_remove_all (rtx->ssrc2_ssrc1_map);
g_hash_table_remove_all (rtx->ssrc1_payload_type_map);
g_hash_table_remove_all (rtx->seqnum_ssrc1_map);
g_hash_table_remove_all (rtx->rtx_payload_type_set);
rtx->num_rtx_requests = 0;
rtx->num_rtx_packets = 0;
rtx->num_rtx_assoc_packets = 0;
g_mutex_unlock (&rtx->lock);
}
static void
gst_rtp_rtx_receive_finalize (GObject * object)
{
GstRtpRtxReceive *rtx = GST_RTP_RTX_RECEIVE (object);
gst_rtp_rtx_receive_reset (rtx);
if (rtx->ssrc2_ssrc1_map) {
g_hash_table_destroy (rtx->ssrc2_ssrc1_map);
rtx->ssrc2_ssrc1_map = NULL;
}
if (rtx->ssrc1_payload_type_map) {
g_hash_table_destroy (rtx->ssrc1_payload_type_map);
rtx->ssrc1_payload_type_map = NULL;
}
if (rtx->seqnum_ssrc1_map) {
g_hash_table_destroy (rtx->seqnum_ssrc1_map);
rtx->seqnum_ssrc1_map = NULL;
}
if (rtx->rtx_payload_type_set) {
g_hash_table_destroy (rtx->rtx_payload_type_set);
rtx->rtx_payload_type_set = NULL;
}
g_mutex_clear (&rtx->lock);
G_OBJECT_CLASS (gst_rtp_rtx_receive_parent_class)->finalize (object);
}
static void
gst_rtp_rtx_receive_init (GstRtpRtxReceive * rtx)
{
GstElementClass *klass = GST_ELEMENT_GET_CLASS (rtx);
rtx->srcpad =
gst_pad_new_from_template (gst_element_class_get_pad_template (klass,
"src"), "src");
GST_PAD_SET_PROXY_CAPS (rtx->srcpad);
GST_PAD_SET_PROXY_ALLOCATION (rtx->srcpad);
gst_pad_set_event_function (rtx->srcpad,
GST_DEBUG_FUNCPTR (gst_rtp_rtx_receive_src_event));
gst_element_add_pad (GST_ELEMENT (rtx), rtx->srcpad);
rtx->sinkpad =
gst_pad_new_from_template (gst_element_class_get_pad_template (klass,
"sink"), "sink");
GST_PAD_SET_PROXY_CAPS (rtx->sinkpad);
GST_PAD_SET_PROXY_ALLOCATION (rtx->sinkpad);
gst_pad_set_chain_function (rtx->sinkpad,
GST_DEBUG_FUNCPTR (gst_rtp_rtx_receive_chain));
gst_element_add_pad (GST_ELEMENT (rtx), rtx->sinkpad);
rtx->ssrc2_ssrc1_map = g_hash_table_new (g_direct_hash, g_direct_equal);
rtx->ssrc1_payload_type_map =
g_hash_table_new (g_direct_hash, g_direct_equal);
rtx->seqnum_ssrc1_map = g_hash_table_new (g_direct_hash, g_direct_equal);
rtx->rtx_payload_type_set = g_hash_table_new (g_direct_hash, g_direct_equal);
g_mutex_init (&rtx->lock);
}
static gboolean
gst_rtp_rtx_receive_src_event (GstPad * pad, GstObject * parent,
GstEvent * event)
{
GstRtpRtxReceive *rtx = GST_RTP_RTX_RECEIVE (parent);
gboolean res;
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_CUSTOM_UPSTREAM:
{
const GstStructure *s = gst_event_get_structure (event);
/* This event usually comes from the downstream gstrtpjitterbuffer */
if (gst_structure_has_name (s, "GstRTPRetransmissionRequest")) {
guint seqnum = 0;
guint ssrc = 0;
gpointer ssrc1 = 0;
gpointer ssrc2 = 0;
/* retrieve seqnum of the packet that need to be restransmisted */
if (!gst_structure_get_uint (s, "seqnum", &seqnum))
seqnum = -1;
/* retrieve ssrc of the packet that need to be restransmisted
* it's usefull when reconstructing the original packet from the rtx packet */
if (!gst_structure_get_uint (s, "ssrc", &ssrc))
ssrc = -1;
GST_DEBUG_OBJECT (rtx,
"request seqnum: %" G_GUINT16_FORMAT ", ssrc: %" G_GUINT32_FORMAT,
seqnum, ssrc);
g_mutex_lock (&rtx->lock);
/* increase number of seen requests for our statistics */
++rtx->num_rtx_requests;
/* First, we lookup in our map to see if we have already associate this
* master stream ssrc with its retransmisted stream.
* Every ssrc are unique so we can use the same hash table
* for both retrieving the ssrc1 from ssrc2 and also ssrc2 from ssrc1
*/
if (g_hash_table_lookup_extended (rtx->ssrc2_ssrc1_map,
GUINT_TO_POINTER (ssrc), NULL, &ssrc2)
&& GPOINTER_TO_UINT (ssrc2) != GPOINTER_TO_UINT (ssrc)) {
GST_DEBUG ("Retransmited stream %" G_GUINT32_FORMAT
" already associated to its master", GPOINTER_TO_UINT (ssrc2));
} else {
/* not already associated but also we have to check that we have not
* already considered this request.
*/
if (g_hash_table_lookup_extended (rtx->seqnum_ssrc1_map,
GUINT_TO_POINTER (seqnum), NULL, &ssrc1)) {
if (GPOINTER_TO_UINT (ssrc1) == ssrc) {
/* do nothing because we have already considered this request
* The jitter may be too impatient of the rtx packet has been
* lost too.
* It does not mean we reject the event, we still want to forward
* the request to the gstrtpsession to be translater into a FB NACK
*/
GST_DEBUG ("Duplicated request seqnum: %" G_GUINT16_FORMAT
", ssrc1: %" G_GUINT32_FORMAT, seqnum, ssrc);
} else {
/* From RFC 4588:
* the receiver MUST NOT have two outstanding requests for the
* same packet sequence number in two different original streams
* before the association is resolved. Otherwise it's impossible
* to associate a rtx stream and its master stream
*/
GST_DEBUG ("reject request for seqnum %" G_GUINT16_FORMAT
"of master stream %" G_GUINT32_FORMAT, seqnum, ssrc);
res = TRUE;
/* remove seqnum in order to reuse the spot */
g_hash_table_remove (rtx->seqnum_ssrc1_map,
GUINT_TO_POINTER (seqnum));
/* do not forward the event as we are rejecting this request */
g_mutex_unlock (&rtx->lock);
gst_event_unref (event);
return res;
}
} else {
/* the request has not been already considered
* insert it for the first time */
GST_DEBUG
("packet number %" G_GUINT16_FORMAT " of master stream %"
G_GUINT32_FORMAT " needs to be retransmited", seqnum, ssrc);
g_hash_table_insert (rtx->seqnum_ssrc1_map,
GUINT_TO_POINTER (seqnum), GUINT_TO_POINTER (ssrc));
}
}
g_mutex_unlock (&rtx->lock);
}
/* Transfer event upstream so that the request can acutally by translated
* through gstrtpsession through the network */
res = gst_pad_event_default (pad, parent, event);
break;
}
default:
res = gst_pad_event_default (pad, parent, event);
break;
}
return res;
}
/* Copy fixed header and extension. Replace current ssrc by ssrc1,
* remove OSN and replace current seq num by OSN.
* Copy memory to avoid to manually copy each rtp buffer field.
*/
static GstBuffer *
_gst_rtp_buffer_new_from_rtx (GstRTPBuffer * rtp, guint32 ssrc1,
guint16 orign_seqnum, guint8 origin_payload_type)
{
GstMemory *mem = NULL;
GstRTPBuffer new_rtp = GST_RTP_BUFFER_INIT;
GstBuffer *new_buffer = gst_buffer_new ();
GstMapInfo map;
guint payload_len = 0;
/* copy fixed header */
mem = gst_memory_copy (rtp->map[0].memory, 0, rtp->size[0]);
gst_buffer_append_memory (new_buffer, mem);
/* copy extension if any */
if (rtp->size[1]) {
mem = gst_memory_copy (rtp->map[1].memory, 0, rtp->size[1]);
gst_buffer_append_memory (new_buffer, mem);
}
/* copy payload and remove OSN */
payload_len = rtp->size[2] - 2;
mem = gst_allocator_alloc (NULL, payload_len, NULL);
gst_memory_map (mem, &map, GST_MAP_WRITE);
if (rtp->size[2])
memcpy (map.data, (guint8 *) rtp->data[2] + 2, payload_len);
gst_memory_unmap (mem, &map);
gst_buffer_append_memory (new_buffer, mem);
/* the sender always constructs rtx packets without padding,
* But the receiver can still receive rtx packets with padding.
* So just copy it.
*/
if (rtp->size[3]) {
guint pad_len = rtp->size[3];
mem = gst_allocator_alloc (NULL, pad_len, NULL);
gst_memory_map (mem, &map, GST_MAP_WRITE);
map.data[pad_len - 1] = pad_len;
gst_memory_unmap (mem, &map);
gst_buffer_append_memory (new_buffer, mem);
}
/* set ssrc and seq num */
gst_rtp_buffer_map (new_buffer, GST_MAP_WRITE, &new_rtp);
gst_rtp_buffer_set_ssrc (&new_rtp, ssrc1);
gst_rtp_buffer_set_seq (&new_rtp, orign_seqnum);
gst_rtp_buffer_set_payload_type (&new_rtp, origin_payload_type);
gst_rtp_buffer_unmap (&new_rtp);
return new_buffer;
}
static GstFlowReturn
gst_rtp_rtx_receive_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
{
GstRtpRtxReceive *rtx = GST_RTP_RTX_RECEIVE (parent);
GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *new_buffer = NULL;
guint32 ssrc = 0;
gpointer ssrc1 = 0;
guint32 ssrc2 = 0;
guint16 seqnum = 0;
guint16 orign_seqnum = 0;
guint8 payload_type = 0;
guint8 origin_payload_type = 0;
gboolean is_rtx = FALSE;
gboolean drop = FALSE;
/* map current rtp packet to parse its header */
gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp);
ssrc = gst_rtp_buffer_get_ssrc (&rtp);
seqnum = gst_rtp_buffer_get_seq (&rtp);
payload_type = gst_rtp_buffer_get_payload_type (&rtp);
/* check if we have a retransmission packet (this information comes from SDP) */
g_mutex_lock (&rtx->lock);
is_rtx =
g_hash_table_lookup_extended (rtx->rtx_payload_type_set,
GUINT_TO_POINTER (payload_type), NULL, NULL);
g_mutex_unlock (&rtx->lock);
if (is_rtx) {
/* read OSN in the rtx payload */
orign_seqnum = GST_READ_UINT16_BE (gst_rtp_buffer_get_payload (&rtp));
}
g_mutex_lock (&rtx->lock);
/* if the current packet is from a retransmission stream */
if (is_rtx) {
/* increase our statistic */
++rtx->num_rtx_packets;
/* first we check if we already have associated this retransmission stream
* to a master stream */
if (g_hash_table_lookup_extended (rtx->ssrc2_ssrc1_map,
GUINT_TO_POINTER (ssrc), NULL, &ssrc1)) {
GST_DEBUG
("packet is from retransmission stream %" G_GUINT32_FORMAT
" already associated to master stream %" G_GUINT32_FORMAT, ssrc,
GPOINTER_TO_UINT (ssrc1));
ssrc2 = ssrc;
/* also retrieve the payload type of the original stream in order to
* reconstruct the packet */
origin_payload_type =
GPOINTER_TO_UINT (g_hash_table_lookup (rtx->ssrc1_payload_type_map,
ssrc1));
} else {
/* the current retransmisted packet has its rtx stream not already
* associated to a master stream, so retrieve it from our request
* history */
if (g_hash_table_lookup_extended (rtx->seqnum_ssrc1_map,
GUINT_TO_POINTER (orign_seqnum), NULL, &ssrc1)) {
GST_DEBUG
("associate retransmisted stream %" G_GUINT32_FORMAT
" to master stream %" G_GUINT32_FORMAT " thanks to packet %"
G_GUINT16_FORMAT "", ssrc, GPOINTER_TO_UINT (ssrc1), orign_seqnum);
ssrc2 = ssrc;
/* free the spot so that this seqnum can be used to do another
* association */
g_hash_table_remove (rtx->seqnum_ssrc1_map,
GUINT_TO_POINTER (orign_seqnum));
/* actually do the association between rtx stream and master stream */
g_hash_table_insert (rtx->ssrc2_ssrc1_map, GUINT_TO_POINTER (ssrc2),
ssrc1);
/* just put a guard */
if (GPOINTER_TO_UINT (ssrc1) == ssrc2)
g_warning
("RTX receiver ssrc2_ssrc1_map bad state, ssrc %" G_GUINT32_FORMAT
" are the same\n", ssrc);
/* also do the association between master stream and rtx stream
* every ssrc are unique so we can use the same hash table
* for both retrieving the ssrc1 from ssrc2 and also ssrc2 from ssrc1
*/
g_hash_table_insert (rtx->ssrc2_ssrc1_map, ssrc1,
GUINT_TO_POINTER (ssrc2));
/* retrieve the original payload type */
origin_payload_type =
GPOINTER_TO_UINT (g_hash_table_lookup (rtx->ssrc1_payload_type_map,
ssrc1));
} else {
/* we are not able to associate this rtx packet with a master stream */
GST_DEBUG
("drop rtx packet because its orign_seqnum %" G_GUINT16_FORMAT
" is not in pending retransmission requests", orign_seqnum);
drop = TRUE;
}
}
} else { /* not rtx */
/* store ssrc -> pt association */
g_hash_table_insert (rtx->ssrc1_payload_type_map, GUINT_TO_POINTER (ssrc),
GUINT_TO_POINTER (payload_type));
}
/* if not dropped the packet was successfully associated */
if (is_rtx && !drop)
++rtx->num_rtx_assoc_packets;
g_mutex_unlock (&rtx->lock);
/* just drop the packet if the association could not have been made */
if (drop) {
gst_rtp_buffer_unmap (&rtp);
gst_buffer_unref (buffer);
return GST_FLOW_OK;
}
/* create the retransmission packet */
if (is_rtx)
new_buffer =
_gst_rtp_buffer_new_from_rtx (&rtp, GPOINTER_TO_UINT (ssrc1),
orign_seqnum, origin_payload_type);
gst_rtp_buffer_unmap (&rtp);
/* push the packet */
if (is_rtx) {
gst_buffer_unref (buffer);
GST_LOG_OBJECT (rtx, "push packet seqnum:%" G_GUINT16_FORMAT
" from a restransmission stream ssrc2:%" G_GUINT32_FORMAT " (src %"
G_GUINT32_FORMAT ")", orign_seqnum, ssrc2, GPOINTER_TO_UINT (ssrc1));
ret = gst_pad_push (rtx->srcpad, new_buffer);
} else {
GST_LOG_OBJECT (rtx, "push packet seqnum:%" G_GUINT16_FORMAT
" from a master stream ssrc: %" G_GUINT32_FORMAT, seqnum, ssrc);
ret = gst_pad_push (rtx->srcpad, buffer);
}
return ret;
}
static void
construct_pt_string (gpointer key, gpointer value, gpointer user_data)
{
GString **str = (GString **) user_data;
if (!(*str)) {
*str = g_string_new (NULL);
g_string_printf (*str, "%d", GPOINTER_TO_UINT (key));
} else {
g_string_append_printf (*str, ":%d", GPOINTER_TO_UINT (key));
}
}
static void
gst_rtp_rtx_receive_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec)
{
GstRtpRtxReceive *rtx = GST_RTP_RTX_RECEIVE (object);
switch (prop_id) {
case PROP_RTX_PAYLOAD_TYPES:{
GString *str = NULL;
g_mutex_lock (&rtx->lock);
g_hash_table_foreach (rtx->rtx_payload_type_set,
(GHFunc) construct_pt_string, &str);
if (str)
g_value_take_string (value, g_string_free (str, FALSE));
g_mutex_unlock (&rtx->lock);
break;
}
case PROP_NUM_RTX_REQUESTS:
g_mutex_lock (&rtx->lock);
g_value_set_uint (value, rtx->num_rtx_requests);
g_mutex_unlock (&rtx->lock);