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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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7d5bb1ea7a
The transport stream only returned the CAPS for the first matching PT entry from the `ptmap`. Other SSRC with the same PT where not included. For a stream which bundled multiple audio streams for instance, only the first SSRC was knowed to the SSRC demux and downstream elements. This commit adds all the `ssrc-` attributes from the matching PT entries. The RTP jitter buffer can now find the CNAME corresponding its SSRC even if it was not the first to be registered for a particular PT. The RTP PT demux removes `ssrc-*` attributes cooresponding to other SSRCs before pushing SSRC specific CAPS to downstream elements. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6119>
407 lines
12 KiB
C
407 lines
12 KiB
C
/* GStreamer
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* Copyright (C) 2017 Matthew Waters <matthew@centricular.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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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#include "transportstream.h"
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#include "transportsendbin.h"
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#include "transportreceivebin.h"
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#include "gstwebrtcbin.h"
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#include "utils.h"
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#include "gst/webrtc/webrtc-priv.h"
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#define GST_CAT_DEFAULT transport_stream_debug
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GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
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#define transport_stream_parent_class parent_class
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G_DEFINE_TYPE_WITH_CODE (TransportStream, transport_stream, GST_TYPE_OBJECT,
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GST_DEBUG_CATEGORY_INIT (GST_CAT_DEFAULT, "webrtctransportstream", 0,
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"webrtctransportstream"););
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enum
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{
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PROP_0,
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PROP_WEBRTC,
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PROP_SESSION_ID,
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PROP_DTLS_CLIENT,
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};
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GstCaps *
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transport_stream_get_caps_for_pt (TransportStream * stream, guint pt)
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{
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GstCaps *ret = NULL;
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GstStructure *ret_s, *item_s;
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guint i, len, si, slen;
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const gchar *field_name;
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len = stream->ptmap->len;
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for (i = 0; i < len; i++) {
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PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
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if (item->pt == pt) {
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if (item->caps) {
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if (ret == NULL) {
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ret = gst_caps_copy (item->caps);
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ret_s = gst_caps_get_structure (ret, 0);
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} else {
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/* Append the "ssrc-*" fields for current PT entry to ret */
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item_s = gst_caps_get_structure (item->caps, 0);
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slen = gst_structure_n_fields (item_s);
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for (si = 0; si < slen; ++si) {
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field_name = gst_structure_nth_field_name (item_s, si);
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if (!g_str_has_prefix (field_name, "ssrc-"))
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continue;
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gst_structure_set (ret_s, field_name, G_TYPE_STRING,
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gst_structure_get_string (item_s, field_name), NULL);
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}
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}
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}
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}
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}
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return ret;
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}
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int
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transport_stream_get_pt (TransportStream * stream, const gchar * encoding_name,
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guint media_idx)
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{
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guint i;
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gint ret = -1;
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for (i = 0; i < stream->ptmap->len; i++) {
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PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
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if (media_idx != -1 && media_idx != item->media_idx)
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continue;
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if (!gst_caps_is_empty (item->caps)) {
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GstStructure *s = gst_caps_get_structure (item->caps, 0);
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if (!g_strcmp0 (gst_structure_get_string (s, "encoding-name"),
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encoding_name)) {
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ret = item->pt;
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break;
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}
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}
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}
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return ret;
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}
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int *
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transport_stream_get_all_pt (TransportStream * stream,
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const gchar * encoding_name, gsize * pt_len)
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{
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guint i;
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gsize ret_i = 0;
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gsize ret_size = 8;
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int *ret = NULL;
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for (i = 0; i < stream->ptmap->len; i++) {
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PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
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if (!gst_caps_is_empty (item->caps)) {
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GstStructure *s = gst_caps_get_structure (item->caps, 0);
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if (!g_strcmp0 (gst_structure_get_string (s, "encoding-name"),
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encoding_name)) {
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if (!ret)
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ret = g_new0 (int, ret_size);
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if (ret_i >= ret_size) {
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ret_size *= 2;
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ret = g_realloc_n (ret, ret_size, sizeof (int));
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}
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ret[ret_i++] = item->pt;
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}
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}
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}
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*pt_len = ret_i;
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return ret;
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}
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static void
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transport_stream_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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TransportStream *stream = TRANSPORT_STREAM (object);
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switch (prop_id) {
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case PROP_WEBRTC:
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gst_object_set_parent (GST_OBJECT (stream), g_value_get_object (value));
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break;
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}
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GST_OBJECT_LOCK (stream);
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switch (prop_id) {
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case PROP_WEBRTC:
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break;
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case PROP_SESSION_ID:
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stream->session_id = g_value_get_uint (value);
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break;
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case PROP_DTLS_CLIENT:
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stream->dtls_client = g_value_get_boolean (value);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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GST_OBJECT_UNLOCK (stream);
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}
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static void
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transport_stream_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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TransportStream *stream = TRANSPORT_STREAM (object);
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GST_OBJECT_LOCK (stream);
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switch (prop_id) {
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case PROP_SESSION_ID:
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g_value_set_uint (value, stream->session_id);
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break;
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case PROP_DTLS_CLIENT:
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g_value_set_boolean (value, stream->dtls_client);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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GST_OBJECT_UNLOCK (stream);
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}
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static void
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transport_stream_dispose (GObject * object)
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{
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TransportStream *stream = TRANSPORT_STREAM (object);
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gst_clear_object (&stream->send_bin);
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gst_clear_object (&stream->receive_bin);
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gst_clear_object (&stream->transport);
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gst_clear_object (&stream->rtxsend);
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gst_clear_object (&stream->rtxreceive);
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gst_clear_object (&stream->reddec);
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g_list_free_full (stream->fecdecs, (GDestroyNotify) gst_object_unref);
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stream->fecdecs = NULL;
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GST_OBJECT_PARENT (object) = NULL;
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G_OBJECT_CLASS (parent_class)->dispose (object);
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}
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static void
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transport_stream_finalize (GObject * object)
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{
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TransportStream *stream = TRANSPORT_STREAM (object);
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g_array_free (stream->ptmap, TRUE);
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g_ptr_array_free (stream->ssrcmap, TRUE);
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gst_clear_object (&stream->rtxsend_stream_id);
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gst_clear_object (&stream->rtxsend_repaired_stream_id);
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gst_clear_object (&stream->rtxreceive_stream_id);
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gst_clear_object (&stream->rtxreceive_repaired_stream_id);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static void
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transport_stream_constructed (GObject * object)
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{
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TransportStream *stream = TRANSPORT_STREAM (object);
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GstWebRTCBin *webrtc;
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GstWebRTCICETransport *ice_trans;
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stream->transport = gst_webrtc_dtls_transport_new (stream->session_id);
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webrtc = GST_WEBRTC_BIN (gst_object_get_parent (GST_OBJECT (object)));
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g_object_bind_property (stream->transport, "client", stream, "dtls-client",
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G_BINDING_BIDIRECTIONAL);
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/* Need to go full Java and have a transport manager?
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* Or make the caller set the ICE transport up? */
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stream->stream = _find_ice_stream_for_session (webrtc, stream->session_id);
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if (stream->stream == NULL) {
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stream->stream = gst_webrtc_ice_add_stream (webrtc->priv->ice,
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stream->session_id);
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_add_ice_stream_item (webrtc, stream->session_id, stream->stream);
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}
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ice_trans =
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gst_webrtc_ice_find_transport (webrtc->priv->ice, stream->stream,
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GST_WEBRTC_ICE_COMPONENT_RTP);
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gst_webrtc_dtls_transport_set_transport (stream->transport, ice_trans);
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gst_object_unref (ice_trans);
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stream->send_bin = g_object_new (transport_send_bin_get_type (), "stream",
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stream, NULL);
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gst_object_ref_sink (stream->send_bin);
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stream->receive_bin = g_object_new (transport_receive_bin_get_type (),
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"stream", stream, NULL);
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gst_object_ref_sink (stream->receive_bin);
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gst_object_unref (webrtc);
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G_OBJECT_CLASS (parent_class)->constructed (object);
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}
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static void
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transport_stream_class_init (TransportStreamClass * klass)
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{
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GObjectClass *gobject_class = (GObjectClass *) klass;
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gobject_class->constructed = transport_stream_constructed;
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gobject_class->get_property = transport_stream_get_property;
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gobject_class->set_property = transport_stream_set_property;
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gobject_class->dispose = transport_stream_dispose;
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gobject_class->finalize = transport_stream_finalize;
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/* some acrobatics are required to set the parent before _constructed()
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* has been called */
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g_object_class_install_property (gobject_class,
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PROP_WEBRTC,
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g_param_spec_object ("webrtc", "Parent webrtcbin",
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"Parent webrtcbin",
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GST_TYPE_WEBRTC_BIN,
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G_PARAM_WRITABLE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class,
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PROP_SESSION_ID,
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g_param_spec_uint ("session-id", "Session ID",
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"Session ID used for this transport",
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0, G_MAXUINT, 0,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class,
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PROP_DTLS_CLIENT,
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g_param_spec_boolean ("dtls-client", "DTLS client",
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"Whether we take the client role in DTLS negotiation",
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FALSE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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}
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static void
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clear_ptmap_item (PtMapItem * item)
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{
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if (item->caps)
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gst_caps_unref (item->caps);
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}
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static SsrcMapItem *
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ssrcmap_item_new (GstWebRTCRTPTransceiverDirection direction, guint32 ssrc,
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guint media_idx)
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{
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SsrcMapItem *ssrc_item = g_new0 (SsrcMapItem, 1);
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ssrc_item->direction = direction;
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ssrc_item->media_idx = media_idx;
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ssrc_item->ssrc = ssrc;
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g_weak_ref_init (&ssrc_item->rtpjitterbuffer, NULL);
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return ssrc_item;
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}
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static void
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ssrcmap_item_free (SsrcMapItem * item)
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{
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g_weak_ref_clear (&item->rtpjitterbuffer);
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g_clear_pointer (&item->mid, g_free);
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g_clear_pointer (&item->rid, g_free);
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g_free (item);
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}
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SsrcMapItem *
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transport_stream_find_ssrc_map_item (TransportStream * stream,
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gconstpointer data, FindSsrcMapFunc func)
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{
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int i;
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for (i = 0; i < stream->ssrcmap->len; i++) {
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SsrcMapItem *item = g_ptr_array_index (stream->ssrcmap, i);
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if (func (item, data))
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return item;
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}
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return NULL;
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}
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void
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transport_stream_filter_ssrc_map_item (TransportStream * stream,
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gconstpointer data, FindSsrcMapFunc func)
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{
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int i;
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for (i = 0; i < stream->ssrcmap->len;) {
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SsrcMapItem *item = g_ptr_array_index (stream->ssrcmap, i);
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if (!func (item, data)) {
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GST_TRACE_OBJECT (stream, "removing ssrc %u", item->ssrc);
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g_ptr_array_remove_index_fast (stream->ssrcmap, i);
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} else {
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i++;
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}
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}
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}
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SsrcMapItem *
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transport_stream_add_ssrc_map_item (TransportStream * stream,
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GstWebRTCRTPTransceiverDirection direction, guint32 ssrc, guint media_idx)
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{
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SsrcMapItem *ret = NULL;
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g_return_val_if_fail (direction ==
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GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY
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|| direction == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY, NULL);
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g_return_val_if_fail (ssrc != 0, NULL);
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GST_INFO_OBJECT (stream, "Adding mapping for rtp session %u media_idx %u "
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"direction %s ssrc %u", stream->session_id, media_idx,
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gst_webrtc_rtp_transceiver_direction_to_string (direction), ssrc);
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/* XXX: duplicates? */
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ret = ssrcmap_item_new (direction, ssrc, media_idx);
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g_ptr_array_add (stream->ssrcmap, ret);
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return ret;
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}
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static void
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transport_stream_init (TransportStream * stream)
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{
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stream->ptmap = g_array_new (FALSE, TRUE, sizeof (PtMapItem));
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g_array_set_clear_func (stream->ptmap, (GDestroyNotify) clear_ptmap_item);
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stream->ssrcmap = g_ptr_array_new_with_free_func (
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(GDestroyNotify) ssrcmap_item_free);
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stream->rtphdrext_id_stream_id = -1;
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stream->rtphdrext_id_repaired_stream_id = -1;
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}
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TransportStream *
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transport_stream_new (GstWebRTCBin * webrtc, guint session_id)
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{
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TransportStream *stream;
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stream = g_object_new (transport_stream_get_type (), "webrtc", webrtc,
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"session-id", session_id, NULL);
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return stream;
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}
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