mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-18 07:47:17 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/subprojects/gst-plugins-good/gst/rtpmanager/gstrtprtxreceive.c

1081 lines
38 KiB
C
Raw Blame History

/* 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
* @title: rtprtxreceive
* @see_also: rtprtxsend, rtpsession, rtpjitterbuffer
*
* rtprtxreceive listens to the retransmission events from the
* downstream rtpjitterbuffer and remembers the SSRC (ssrc1) of the stream and
* the sequence number that was requested. When it receives a packet with
* a sequence number equal to one of the ones stored and with a different SSRC,
* it identifies the new SSRC (ssrc2) as the retransmission stream of ssrc1.
* From this point on, it replaces ssrc2 with ssrc1 in all packets of the
* ssrc2 stream and flags them as retransmissions, so that rtpjitterbuffer
* can reconstruct the original stream.
*
* This algorithm is implemented as specified in RFC 4588.
*
* This element is meant to be used with rtprtxsend on the sender side.
* See #GstRtpRtxSend
*
* Below you can see some examples that illustrate how rtprtxreceive and
* rtprtxsend fit among the other rtp elements and how they work internally.
* Normally, hoewever, you should avoid using such pipelines and use
* rtpbin instead, with its #GstRtpBin::request-aux-sender and
* #GstRtpBin::request-aux-receiver signals. See #GstRtpBin.
*
* ## Example pipelines
*
* |[
* gst-launch-1.0 rtpsession name=rtpsession rtp-profile=avpf \
* audiotestsrc is-live=true ! opusenc ! rtpopuspay pt=96 ! \
* rtprtxsend payload-type-map="application/x-rtp-pt-map,96=(uint)97" ! \
* rtpsession.send_rtp_sink \
* rtpsession.send_rtp_src ! identity drop-probability=0.01 ! \
* 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 rtp-profile=avpf \
* udpsrc port=5000 caps="application/x-rtp,media=(string)audio,clock-rate=(int)48000,encoding-name=(string)OPUS,payload=(int)96" ! \
* rtpsession.recv_rtp_sink \
* rtpsession.recv_rtp_src ! \
* rtprtxreceive payload-type-map="application/x-rtp-pt-map,96=(uint)97" ! \
* rtpssrcdemux ! rtpjitterbuffer do-retransmission=true ! \
* rtpopusdepay ! opusdec ! audioconvert ! audioresample ! autoaudiosink \
* rtpsession.send_rtcp_src ! \
* udpsink host="127.0.0.1" port=5001 sync=false async=false \
* udpsrc port=5002 ! rtpsession.recv_rtcp_sink
* ]|
* Receive audio stream from port 5000 (5001 and 5002 are just the rtcp
* link with the sender)
*
* In this example we can see a simple streaming of an OPUS stream with some
* of the packets being artificially dropped by the identity element.
* Thanks to retransmission, you should still hear a clear sound when setting
* drop-probability to something greater than 0.
*
* Internally, the rtpjitterbuffer will generate a custom upstream event,
* GstRTPRetransmissionRequest, when it detects that one packet is missing.
* Then this request is translated to a FB NACK in the rtcp link by rtpsession.
* Finally the rtpsession of the sender side will re-convert it in a
* GstRTPRetransmissionRequest that will be handled by rtprtxsend. rtprtxsend
* will then re-send the missing packet with a new srrc and a different payload
* type (here, 97), but with the same original sequence number. On the receiver
* side, rtprtxreceive will associate this new stream with the original and
* forward the retransmission packets to rtpjitterbuffer with the original
* ssrc and payload type.
*
* |[
* gst-launch-1.0 rtpsession name=rtpsession rtp-profile=avpf \
* audiotestsrc is-live=true ! opusenc ! rtpopuspay pt=97 seqnum-offset=1 ! \
* rtprtxsend payload-type-map="application/x-rtp-pt-map,97=(uint)99" ! \
* funnel name=f ! rtpsession.send_rtp_sink \
* audiotestsrc freq=660.0 is-live=true ! opusenc ! \
* rtpopuspay pt=97 seqnum-offset=100 ! \
* rtprtxsend payload-type-map="application/x-rtp-pt-map,97=(uint)99" ! \
* f. \
* rtpsession.send_rtp_src ! identity drop-probability=0.01 ! \
* 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 two audio streams to port 5000.
* |[
* gst-launch-1.0 rtpsession name=rtpsession rtp-profile=avpf \
* udpsrc port=5000 caps="application/x-rtp,media=(string)audio,clock-rate=(int)48000,encoding-name=(string)OPUS,payload=(int)97" ! \
* rtpsession.recv_rtp_sink \
* rtpsession.recv_rtp_src ! \
* rtprtxreceive payload-type-map="application/x-rtp-pt-map,97=(uint)99" ! \
* rtpssrcdemux name=demux \
* demux. ! queue ! rtpjitterbuffer do-retransmission=true ! rtpopusdepay ! \
* opusdec ! audioconvert ! autoaudiosink \
* demux. ! queue ! rtpjitterbuffer do-retransmission=true ! rtpopusdepay ! \
* opusdec ! audioconvert ! autoaudiosink \
* udpsrc port=5002 ! rtpsession.recv_rtcp_sink \
* rtpsession.send_rtcp_src ! udpsink host="127.0.0.1" port=5001 \
* sync=false async=false
* ]|
* Receive two audio streams from port 5000.
*
* In this example we are streaming two streams of the same type through the
* same port. They, however, are using a different SSRC (ssrc is randomly
* generated on each payloader - rtpopuspay in this example), so they can be
* identified and demultiplexed by rtpssrcdemux on the receiver side. This is
* an example of SSRC-multiplexing.
*
* It is important here to use a different starting sequence number
* (seqnum-offset), since this is the only means of identification that
* rtprtxreceive uses the very first time to identify retransmission streams.
* It is an error, according to RFC4588 to have two retransmission requests for
* packets belonging to two different streams but with the same sequence number.
* Note that the default seqnum-offset value (-1, which means random) would
* work just fine, but it is overridden here for illustration purposes.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst.h>
#include <gst/rtp/rtp.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_SSRC_MAP,
PROP_PAYLOAD_TYPE_MAP,
PROP_NUM_RTX_REQUESTS,
PROP_NUM_RTX_PACKETS,
PROP_NUM_RTX_ASSOC_PACKETS
};
enum
{
SIGNAL_0,
SIGNAL_ADD_EXTENSION,
SIGNAL_CLEAR_EXTENSIONS,
LAST_SIGNAL
};
static guint gst_rtp_rtx_receive_signals[LAST_SIGNAL] = { 0, };
#define RTPHDREXT_STREAM_ID GST_RTP_HDREXT_BASE "sdes:rtp-stream-id"
#define RTPHDREXT_REPAIRED_STREAM_ID GST_RTP_HDREXT_BASE "sdes:repaired-rtp-stream-id"
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_WITH_CODE (GstRtpRtxReceive, gst_rtp_rtx_receive,
GST_TYPE_ELEMENT, GST_DEBUG_CATEGORY_INIT (gst_rtp_rtx_receive_debug,
"rtprtxreceive", 0, "rtp retransmission receiver"));
GST_ELEMENT_REGISTER_DEFINE (rtprtxreceive, "rtprtxreceive", GST_RANK_NONE,
GST_TYPE_RTP_RTX_RECEIVE);
static void
gst_rtp_rtx_receive_add_extension (GstRtpRtxReceive * rtx,
GstRTPHeaderExtension * ext)
{
g_return_if_fail (GST_IS_RTP_HEADER_EXTENSION (ext));
g_return_if_fail (gst_rtp_header_extension_get_id (ext) > 0);
GST_OBJECT_LOCK (rtx);
if (g_strcmp0 (gst_rtp_header_extension_get_uri (ext),
RTPHDREXT_STREAM_ID) == 0) {
gst_clear_object (&rtx->rid_stream);
rtx->rid_stream = gst_object_ref (ext);
} else if (g_strcmp0 (gst_rtp_header_extension_get_uri (ext),
RTPHDREXT_REPAIRED_STREAM_ID) == 0) {
gst_clear_object (&rtx->rid_repaired);
rtx->rid_repaired = gst_object_ref (ext);
} else {
g_warning ("rtprtxsend (%s) doesn't know how to deal with the "
"RTP Header Extension with URI \'%s\'", GST_OBJECT_NAME (rtx),
gst_rtp_header_extension_get_uri (ext));
}
/* XXX: check for other duplicate ids? */
GST_OBJECT_UNLOCK (rtx);
}
static void
gst_rtp_rtx_receive_clear_extensions (GstRtpRtxReceive * rtx)
{
GST_OBJECT_LOCK (rtx);
gst_clear_object (&rtx->rid_stream);
gst_clear_object (&rtx->rid_repaired);
GST_OBJECT_UNLOCK (rtx);
}
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;
/**
* GstRtpRtxReceive:ssrc-map:
*
* Map of SSRCs to their retransmission SSRCs for SSRC-multiplexed mode.
*
* If an application know this information already (WebRTC signals this
* in their SDP), it can allow the rtxreceive element to know a packet
* is a "valid" RTX packet even if it has not been requested.
*
* Since: 1.22
*/
g_object_class_install_property (gobject_class, PROP_SSRC_MAP,
g_param_spec_boxed ("ssrc-map", "SSRC Map",
"Map of SSRCs to their retransmission SSRCs for SSRC-multiplexed mode",
GST_TYPE_STRUCTURE, G_PARAM_WRITABLE | G_PARAM_STATIC_STRINGS));
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));
/**
* rtprtxreceive::add-extension:
*
* Add @ext as an extension for writing part of an RTP header extension onto
* outgoing RTP packets. Currently only supports using the following
* extension URIs. All other RTP header extensions are copied as-is.
* - "urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id": will be removed
* - "urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id": will be
* written instead of the "rtp-stream-id" header extension.
*
* Since: 1.22
*/
gst_rtp_rtx_receive_signals[SIGNAL_ADD_EXTENSION] =
g_signal_new_class_handler ("add-extension", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
G_CALLBACK (gst_rtp_rtx_receive_add_extension), NULL, NULL, NULL,
G_TYPE_NONE, 1, GST_TYPE_RTP_HEADER_EXTENSION);
/**
* rtprtxreceive::clear-extensions:
* @object: the #GstRTPBasePayload
*
* Clear all RTP header extensions used by rtprtxreceive.
*
* Since: 1.22
*/
gst_rtp_rtx_receive_signals[SIGNAL_CLEAR_EXTENSIONS] =
g_signal_new_class_handler ("clear-extensions", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
G_CALLBACK (gst_rtp_rtx_receive_clear_extensions), NULL, NULL, NULL,
G_TYPE_NONE, 0);
gst_element_class_add_static_pad_template (gstelement_class, &src_factory);
gst_element_class_add_static_pad_template (gstelement_class, &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_CAST (object);
g_hash_table_unref (rtx->ssrc2_ssrc1_map);
if (rtx->external_ssrc_map)
gst_structure_free (rtx->external_ssrc_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);
gst_clear_object (&rtx->rid_stream);
gst_clear_object (&rtx->rid_repaired);
gst_clear_buffer (&rtx->dummy_writable);
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_new (SsrcAssoc, 1);
assoc->ssrc = ssrc;
assoc->time = time;
return assoc;
}
static void
ssrc_assoc_free (SsrcAssoc * assoc)
{
g_free (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);
rtx->dummy_writable = gst_buffer_new ();
}
static gboolean
gst_rtp_rtx_receive_src_event (GstPad * pad, GstObject * parent,
GstEvent * event)
{
GstRtpRtxReceive *rtx = GST_RTP_RTX_RECEIVE_CAST (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 retransmitted */
if (!gst_structure_get_uint (s, "seqnum", &seqnum))
seqnum = -1;
/* retrieve ssrc of the packet that need to be retransmitted
* it's useful when reconstructing the original packet from the rtx packet */
if (!gst_structure_get_uint (s, "ssrc", &ssrc))
ssrc = -1;
GST_DEBUG_OBJECT (rtx, "got rtx request for seqnum: %u, ssrc: %X",
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 retransmitted 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_TRACE_OBJECT (rtx, "Retransmitted stream %X already associated "
"to its master, %X", GPOINTER_TO_UINT (ssrc2), ssrc);
} 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) {
/* same seqnum, same 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 translator into a FB NACK
*/
GST_LOG_OBJECT (rtx, "Duplicate request: seqnum: %u, ssrc: %X",
seqnum, ssrc);
} else {
/* same seqnum, different ssrc */
/* 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_INFO_OBJECT (rtx, "rejecting request for seqnum %u"
" of master stream %X; there is already a pending request "
"for the same seqnum on ssrc %X that has not expired",
seqnum, ssrc, assoc->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 */
g_hash_table_insert (rtx->seqnum_ssrc1_map,
GUINT_TO_POINTER (seqnum),
ssrc_assoc_new (ssrc, rtx->last_time));
}
}
GST_DEBUG_OBJECT (rtx, "packet number %u of master stream %X"
" needs to be retransmitted", seqnum, ssrc);
GST_OBJECT_UNLOCK (rtx);
}
/* Transfer event upstream so that the request can actually 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;
}
static GstMemory *
rewrite_header_extensions (GstRtpRtxReceive * rtx, GstRTPBuffer * rtp)
{
gsize out_size = rtp->size[1] + 32;
guint16 bit_pattern;
guint8 *pdata;
guint wordlen;
GstMemory *mem;
GstMapInfo map;
mem = gst_allocator_alloc (NULL, out_size, NULL);
gst_memory_map (mem, &map, GST_MAP_READWRITE);
if (gst_rtp_buffer_get_extension_data (rtp, &bit_pattern, (gpointer) & pdata,
&wordlen)) {
GstRTPHeaderExtensionFlags ext_flags = 0;
gsize bytelen = wordlen * 4;
guint hdr_unit_bytes;
gsize read_offset = 0, write_offset = 4;
if (bit_pattern == 0xBEDE) {
/* one byte extensions */
hdr_unit_bytes = 1;
ext_flags |= GST_RTP_HEADER_EXTENSION_ONE_BYTE;
} else if (bit_pattern >> 4 == 0x100) {
/* two byte extensions */
hdr_unit_bytes = 2;
ext_flags |= GST_RTP_HEADER_EXTENSION_TWO_BYTE;
} else {
GST_DEBUG_OBJECT (rtx, "unknown extension bit pattern 0x%02x%02x",
bit_pattern >> 8, bit_pattern & 0xff);
goto copy_as_is;
}
GST_WRITE_UINT16_BE (map.data, bit_pattern);
while (TRUE) {
guint8 read_id, read_len;
if (read_offset + hdr_unit_bytes >= bytelen)
/* not enough remaning data */
break;
if (ext_flags & GST_RTP_HEADER_EXTENSION_ONE_BYTE) {
read_id = GST_READ_UINT8 (pdata + read_offset) >> 4;
read_len = (GST_READ_UINT8 (pdata + read_offset) & 0x0F) + 1;
read_offset += 1;
if (read_id == 0)
/* padding */
continue;
if (read_id == 15)
/* special id for possible future expansion */
break;
} else {
read_id = GST_READ_UINT8 (pdata + read_offset);
read_offset += 1;
if (read_id == 0)
/* padding */
continue;
read_len = GST_READ_UINT8 (pdata + read_offset);
read_offset += 1;
}
GST_TRACE_OBJECT (rtx, "found rtp header extension with id %u and "
"length %u", read_id, read_len);
/* Ignore extension headers where the size does not fit */
if (read_offset + read_len > bytelen) {
GST_WARNING_OBJECT (rtx, "Extension length extends past the "
"size of the extension data");
break;
}
/* rewrite the rtp-stream-id into a repaired-stream-id */
if (rtx->rid_stream
&& read_id == gst_rtp_header_extension_get_id (rtx->rid_repaired)) {
if (!gst_rtp_header_extension_read (rtx->rid_repaired, ext_flags,
&pdata[read_offset], read_len, rtx->dummy_writable)) {
GST_WARNING_OBJECT (rtx, "RTP header extension (%s) could "
"not read payloaded data", GST_OBJECT_NAME (rtx->rid_stream));
goto copy_as_is;
}
if (rtx->rid_repaired) {
guint8 write_id = gst_rtp_header_extension_get_id (rtx->rid_stream);
gsize written;
char *rid;
g_object_get (rtx->rid_repaired, "rid", &rid, NULL);
g_object_set (rtx->rid_stream, "rid", rid, NULL);
g_clear_pointer (&rid, g_free);
written =
gst_rtp_header_extension_write (rtx->rid_stream, rtp->buffer,
ext_flags, rtx->dummy_writable,
&map.data[write_offset + hdr_unit_bytes],
map.size - write_offset - hdr_unit_bytes);
GST_TRACE_OBJECT (rtx->rid_repaired, "wrote %" G_GSIZE_FORMAT,
written);
if (written <= 0) {
GST_WARNING_OBJECT (rtx, "Failed to rewrite RID for RTX");
goto copy_as_is;
} else {
if (ext_flags & GST_RTP_HEADER_EXTENSION_ONE_BYTE) {
map.data[write_offset] =
((write_id & 0x0F) << 4) | ((written - 1) & 0x0F);
} else if (ext_flags & GST_RTP_HEADER_EXTENSION_TWO_BYTE) {
map.data[write_offset] = write_id & 0xFF;
map.data[write_offset + 1] = written & 0xFF;
} else {
g_assert_not_reached ();
goto copy_as_is;
}
write_offset += written + hdr_unit_bytes;
}
}
} else {
/* TODO: may need to write mid at different times to the original
* buffer to account for the difference in timing of acknowledgement
* of the rtx ssrc from the original ssrc. This may add extra data to
* the header extension space that needs to be accounted for.
*/
memcpy (&map.data[write_offset],
&pdata[read_offset - hdr_unit_bytes], read_len + hdr_unit_bytes);
write_offset += read_len + hdr_unit_bytes;
}
read_offset += read_len;
}
/* subtract the ext header */
wordlen = write_offset / 4 + ((write_offset % 4) ? 1 : 0);
/* wordlen in the ext data doesn't include the 4-byte header */
GST_WRITE_UINT16_BE (map.data + 2, wordlen - 1);
if (wordlen * 4 > write_offset)
memset (&map.data[write_offset], 0, wordlen * 4 - write_offset);
GST_MEMDUMP_OBJECT (rtx, "generated ext data", map.data, wordlen * 4);
} else {
copy_as_is:
wordlen = rtp->size[1] / 4;
memcpy (map.data, rtp->data[1], rtp->size[1]);
GST_LOG_OBJECT (rtx, "copying data as-is");
}
gst_memory_unmap (mem, &map);
gst_memory_resize (mem, 0, wordlen * 4);
return mem;
}
/* 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 (GstRtpRtxReceive * 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 = rewrite_header_extensions (rtx, rtp);
gst_buffer_append_memory (new_buffer, mem);
}
/* copy payload and remove OSN */
g_assert_cmpint (rtp->size[2], >, 1);
payload_len = rtp->size[2] - 2;
mem = gst_allocator_alloc (NULL, payload_len, NULL);
gst_memory_map (mem, &map, GST_MAP_WRITE);
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);
GST_BUFFER_FLAG_SET (new_buffer, GST_RTP_BUFFER_FLAG_RETRANSMISSION);
return new_buffer;
}
static GstFlowReturn
gst_rtp_rtx_receive_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
{
GstRtpRtxReceive *rtx = GST_RTP_RTX_RECEIVE_CAST (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;
gpointer payload = NULL;
guint8 origin_payload_type = 0;
gboolean is_rtx;
gboolean drop = FALSE;
if (rtx->rtx_pt_map_structure == NULL)
goto no_map;
/* map current rtp packet to parse its header */
if (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp))
goto invalid_buffer;
GST_MEMDUMP_OBJECT (rtx, "rtp header", rtp.map[0].data, rtp.map[0].size);
GST_MEMDUMP_OBJECT (rtx, "rtp ext", rtp.map[1].data, rtp.map[1].size);
GST_MEMDUMP_OBJECT (rtx, "rtp payload", rtp.map[2].data, rtp.map[2].size);
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);
is_rtx =
g_hash_table_lookup_extended (rtx->rtx_pt_map,
GUINT_TO_POINTER (payload_type), NULL, NULL);
if (is_rtx) {
payload = gst_rtp_buffer_get_payload (&rtp);
if (!payload || gst_rtp_buffer_get_payload_len (&rtp) < 2) {
GST_OBJECT_UNLOCK (rtx);
gst_rtp_buffer_unmap (&rtp);
goto invalid_buffer;
}
}
rtx->last_time = GST_BUFFER_PTS (buffer);
if (g_hash_table_size (rtx->seqnum_ssrc1_map) > 0) {
GHashTableIter iter;
gpointer key, value;
g_hash_table_iter_init (&iter, rtx->seqnum_ssrc1_map);
while (g_hash_table_iter_next (&iter, &key, &value)) {
SsrcAssoc *assoc = value;
/* remove association request if it is too old */
if (GST_CLOCK_TIME_IS_VALID (rtx->last_time) &&
GST_CLOCK_TIME_IS_VALID (assoc->time) &&
assoc->time + ASSOC_TIMEOUT < rtx->last_time) {
g_hash_table_iter_remove (&iter);
}
}
}
/* if the current packet is from a retransmission stream */
if (is_rtx) {
/* increase our statistic */
++rtx->num_rtx_packets;
/* check if there enough data to read OSN from the paylaod,
we need at least two bytes
*/
if (gst_rtp_buffer_get_payload_len (&rtp) > 1) {
/* 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)));
GST_DEBUG_OBJECT (rtx, "Got rtx packet: rtx seqnum %u, rtx ssrc %X, "
"rtx pt %u, orig seqnum %u, orig pt %u", seqnum, ssrc, payload_type,
orign_seqnum, origin_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_TRACE_OBJECT (rtx,
"packet is from retransmission stream %X already associated to "
"master stream %X", ssrc, GPOINTER_TO_UINT (ssrc1));
ssrc2 = ssrc;
} else {
SsrcAssoc *assoc;
/* the current retransmitted 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_LOG_OBJECT (rtx,
"associating retransmitted stream %X to master stream %X thanks "
"to rtx packet %u (orig seqnum %u)", ssrc, assoc->ssrc, seqnum,
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, "
"master and rtx SSRCs are the same (%X)\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_INFO_OBJECT (rtx,
"dropping rtx packet %u because its orig seqnum (%u) is not in our"
" pending retransmission requests", seqnum, orign_seqnum);
drop = TRUE;
}
}
} else {
/* the rtx packet is empty */
GST_DEBUG_OBJECT (rtx, "drop rtx packet because it is empty");
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 (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, "pushing packet seqnum:%u from restransmission "
"stream ssrc: %X (master ssrc %X)", orign_seqnum, ssrc2,
GPOINTER_TO_UINT (ssrc1));
ret = gst_pad_push (rtx->srcpad, new_buffer);
} else {
GST_TRACE_OBJECT (rtx, "pushing packet seqnum:%u from master stream "
"ssrc: %X", seqnum, ssrc);
ret = gst_pad_push (rtx->srcpad, buffer);
}
return ret;
no_map:
{
GST_DEBUG_OBJECT (pad, "No map set, passthrough");
return gst_pad_push (rtx->srcpad, buffer);
}
invalid_buffer:
{
GST_INFO_OBJECT (pad, "Received invalid RTP payload, dropping");
gst_buffer_unref (buffer);
return GST_FLOW_OK;
}
}
static void
gst_rtp_rtx_receive_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec)
{
GstRtpRtxReceive *rtx = GST_RTP_RTX_RECEIVE_CAST (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_CAST (object);
switch (prop_id) {
case PROP_SSRC_MAP:
GST_OBJECT_LOCK (rtx);
if (rtx->external_ssrc_map)
gst_structure_free (rtx->external_ssrc_map);
rtx->external_ssrc_map = g_value_dup_boxed (value);
g_hash_table_remove_all (rtx->ssrc2_ssrc1_map);
gst_structure_foreach (rtx->external_ssrc_map,
structure_to_hash_table_inv, rtx->ssrc2_ssrc1_map);
GST_OBJECT_UNLOCK (rtx);
break;
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_CAST (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;
}
Reference in a new issue View git blame Copy permalink