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/* 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>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst.h>
#include <gst/rtp/gstrtpbuffer.h>
#include <string.h>
#include <stdlib.h>
#include "gstrtprtxreceive.h"
#define ASSOC_TIMEOUT (GST_SECOND)
GST_DEBUG_CATEGORY_STATIC (gst_rtp_rtx_receive_debug);
#define GST_CAT_DEFAULT gst_rtp_rtx_receive_debug
enum
{
PROP_0,
PROP_PAYLOAD_TYPE_MAP,
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_PAYLOAD_TYPE_MAP,
g_param_spec_boxed ("payload-type-map", "Payload Type Map",
"Map of original payload types to their retransmission payload types",
GST_TYPE_STRUCTURE, 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)
{
GST_OBJECT_LOCK (rtx);
g_hash_table_remove_all (rtx->ssrc2_ssrc1_map);
g_hash_table_remove_all (rtx->seqnum_ssrc1_map);
rtx->num_rtx_requests = 0;
rtx->num_rtx_packets = 0;
rtx->num_rtx_assoc_packets = 0;
GST_OBJECT_UNLOCK (rtx);
}
static void
gst_rtp_rtx_receive_finalize (GObject * object)
{
GstRtpRtxReceive *rtx = GST_RTP_RTX_RECEIVE (object);
g_hash_table_unref (rtx->ssrc2_ssrc1_map);
g_hash_table_unref (rtx->seqnum_ssrc1_map);
g_hash_table_unref (rtx->rtx_pt_map);
if (rtx->rtx_pt_map_structure)
gst_structure_free (rtx->rtx_pt_map_structure);
G_OBJECT_CLASS (gst_rtp_rtx_receive_parent_class)->finalize (object);
}
typedef struct
{
guint32 ssrc;
GstClockTime time;
} SsrcAssoc;
static SsrcAssoc *
ssrc_assoc_new (guint32 ssrc, GstClockTime time)
{
SsrcAssoc *assoc = g_slice_new (SsrcAssoc);
assoc->ssrc = ssrc;
assoc->time = time;
return assoc;
}
static void
ssrc_assoc_free (SsrcAssoc * assoc)
{
g_slice_free (SsrcAssoc, assoc);
}
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->seqnum_ssrc1_map = g_hash_table_new_full (g_direct_hash, g_direct_equal,
NULL, (GDestroyNotify) ssrc_assoc_free);
rtx->rtx_pt_map = g_hash_table_new (g_direct_hash, g_direct_equal);
}
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 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_GUINT32_FORMAT ", ssrc: %" G_GUINT32_FORMAT,
seqnum, ssrc);
GST_OBJECT_LOCK (rtx);
/* 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_OBJECT (rtx, "Retransmited stream %" G_GUINT32_FORMAT
" already associated to its master", GPOINTER_TO_UINT (ssrc2));
} else {
SsrcAssoc *assoc;
/* 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, (gpointer *) & assoc)) {
if (assoc->ssrc == 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_OBJECT (rtx, "Duplicated request seqnum: %"
G_GUINT32_FORMAT ", ssrc1: %" G_GUINT32_FORMAT, seqnum, ssrc);
} else {
/* If the association attempt is larger than ASSOC_TIMEOUT,
* then we give up on it, and try this one.
*/
if (!GST_CLOCK_TIME_IS_VALID (rtx->last_time) ||
!GST_CLOCK_TIME_IS_VALID (assoc->time) ||
assoc->time + ASSOC_TIMEOUT < rtx->last_time) {
/* 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
*/
/* remove seqnum in order to reuse the spot */
g_hash_table_remove (rtx->seqnum_ssrc1_map,
GUINT_TO_POINTER (seqnum));
goto retransmit;
} else {
GST_DEBUG_OBJECT (rtx,
"reject request for seqnum %" G_GUINT32_FORMAT
" of master stream %" G_GUINT32_FORMAT, seqnum, ssrc);
/* do not forward the event as we are rejecting this request */
GST_OBJECT_UNLOCK (rtx);
gst_event_unref (event);
return TRUE;
}
}
} else {
retransmit:
/* the request has not been already considered
* insert it for the first time */
GST_DEBUG_OBJECT (rtx,
"packet number %" G_GUINT32_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),
ssrc_assoc_new (ssrc, rtx->last_time));
}
}
GST_OBJECT_UNLOCK (rtx);
}
/* 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,
(guint8 *) rtp->data[0] - rtp->map[0].data, 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,
(guint8 *) rtp->data[1] - rtp->map[1].data, 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);
gst_buffer_copy_into (new_buffer, rtp->buffer,
GST_BUFFER_COPY_FLAGS | GST_BUFFER_COPY_TIMESTAMPS, 0, -1);
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) */
GST_OBJECT_LOCK (rtx);
rtx->last_time = GST_BUFFER_PTS (buffer);
is_rtx =
g_hash_table_lookup_extended (rtx->rtx_pt_map,
GUINT_TO_POINTER (payload_type), NULL, NULL);
/* if the current packet is from a retransmission stream */
if (is_rtx) {
/* increase our statistic */
++rtx->num_rtx_packets;
/* read OSN in the rtx payload */
orign_seqnum = GST_READ_UINT16_BE (gst_rtp_buffer_get_payload (&rtp));
origin_payload_type =
GPOINTER_TO_UINT (g_hash_table_lookup (rtx->rtx_pt_map,
GUINT_TO_POINTER (payload_type)));
/* 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_OBJECT (rtx,
"packet is from retransmission stream %" G_GUINT32_FORMAT
" already associated to master stream %" G_GUINT32_FORMAT, ssrc,
GPOINTER_TO_UINT (ssrc1));
ssrc2 = ssrc;
} else {
SsrcAssoc *assoc;
/* 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, (gpointer *) & assoc)) {
GST_DEBUG_OBJECT (rtx,
"associate retransmisted stream %" G_GUINT32_FORMAT
" to master stream %" G_GUINT32_FORMAT " thanks to packet %"
G_GUINT16_FORMAT "", ssrc, assoc->ssrc, orign_seqnum);
ssrc1 = GUINT_TO_POINTER (assoc->ssrc);
ssrc2 = ssrc;
/* just put a guard */
if (GPOINTER_TO_UINT (ssrc1) == ssrc2)
GST_WARNING_OBJECT (rtx, "RTX receiver ssrc2_ssrc1_map bad state, "
"ssrc %" G_GUINT32_FORMAT " are the same\n", 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);
/* 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));
} else {
/* we are not able to associate this rtx packet with a master stream */
GST_DEBUG_OBJECT (rtx,
"drop rtx packet because its orign_seqnum %" G_GUINT16_FORMAT
" is not in pending retransmission requests", orign_seqnum);
drop = TRUE;
}
}
}
/* if not dropped the packet was successfully associated */
if (is_rtx && !drop)
++rtx->num_rtx_assoc_packets;
GST_OBJECT_UNLOCK (rtx);
/* 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
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_PAYLOAD_TYPE_MAP:
GST_OBJECT_LOCK (rtx);
g_value_set_boxed (value, rtx->rtx_pt_map_structure);
GST_OBJECT_UNLOCK (rtx);
break;
case PROP_NUM_RTX_REQUESTS:
GST_OBJECT_LOCK (rtx);
g_value_set_uint (value, rtx->num_rtx_requests);
GST_OBJECT_UNLOCK (rtx);
break;
case PROP_NUM_RTX_PACKETS:
GST_OBJECT_LOCK (rtx);
g_value_set_uint (value, rtx->num_rtx_packets);
GST_OBJECT_UNLOCK (rtx);
break;
case PROP_NUM_RTX_ASSOC_PACKETS:
GST_OBJECT_LOCK (rtx);
g_value_set_uint (value, rtx->num_rtx_assoc_packets);
GST_OBJECT_UNLOCK (rtx);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static gboolean
structure_to_hash_table_inv (GQuark field_id, const GValue * value,
gpointer hash)
{
const gchar *field_str;
guint field_uint;
guint value_uint;
field_str = g_quark_to_string (field_id);
field_uint = atoi (field_str);
value_uint = g_value_get_uint (value);
g_hash_table_insert ((GHashTable *) hash, GUINT_TO_POINTER (value_uint),
GUINT_TO_POINTER (field_uint));
return TRUE;
}
static void
gst_rtp_rtx_receive_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec)
{
GstRtpRtxReceive *rtx = GST_RTP_RTX_RECEIVE (object);
switch (prop_id) {
case PROP_PAYLOAD_TYPE_MAP:
GST_OBJECT_LOCK (rtx);
if (rtx->rtx_pt_map_structure)
gst_structure_free (rtx->rtx_pt_map_structure);
rtx->rtx_pt_map_structure = g_value_dup_boxed (value);
g_hash_table_remove_all (rtx->rtx_pt_map);
gst_structure_foreach (rtx->rtx_pt_map_structure,
structure_to_hash_table_inv, rtx->rtx_pt_map);
GST_OBJECT_UNLOCK (rtx);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstStateChangeReturn
gst_rtp_rtx_receive_change_state (GstElement * element,
GstStateChange transition)
{
GstStateChangeReturn ret;
GstRtpRtxReceive *rtx;
rtx = GST_RTP_RTX_RECEIVE (element);
switch (transition) {
default:
break;
}
ret =
GST_ELEMENT_CLASS (gst_rtp_rtx_receive_parent_class)->change_state
(element, transition);
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_rtp_rtx_receive_reset (rtx);
break;
default:
break;
}
return ret;
}
gboolean
gst_rtp_rtx_receive_plugin_init (GstPlugin * plugin)
{
GST_DEBUG_CATEGORY_INIT (gst_rtp_rtx_receive_debug, "rtprtxreceive", 0,
"rtp retransmission receiver");
return gst_element_register (plugin, "rtprtxreceive", GST_RANK_NONE,
GST_TYPE_RTP_RTX_RECEIVE);
}