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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 17:32:43 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 17:32:43 +0000
commit6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch)
treea68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/api/call/transport.h
parentInitial commit. (diff)
downloadthunderbird-upstream.tar.xz
thunderbird-upstream.zip
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/api/call/transport.h')
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diff --git a/third_party/libwebrtc/api/call/transport.h b/third_party/libwebrtc/api/call/transport.h
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+/*
+ * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_CALL_TRANSPORT_H_
+#define API_CALL_TRANSPORT_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "api/ref_counted_base.h"
+#include "api/scoped_refptr.h"
+
+namespace webrtc {
+
+// TODO(holmer): Look into unifying this with the PacketOptions in
+// asyncpacketsocket.h.
+struct PacketOptions {
+ PacketOptions();
+ PacketOptions(const PacketOptions&);
+ ~PacketOptions();
+
+ // A 16 bits positive id. Negative ids are invalid and should be interpreted
+ // as packet_id not being set.
+ int packet_id = -1;
+ // Additional data bound to the RTP packet for use in application code,
+ // outside of WebRTC.
+ rtc::scoped_refptr<rtc::RefCountedBase> additional_data;
+ // Whether this is a retransmission of an earlier packet.
+ bool is_retransmit = false;
+ bool included_in_feedback = false;
+ bool included_in_allocation = false;
+};
+
+class Transport {
+ public:
+ virtual bool SendRtp(const uint8_t* packet,
+ size_t length,
+ const PacketOptions& options) = 0;
+ virtual bool SendRtcp(const uint8_t* packet, size_t length) = 0;
+
+ protected:
+ virtual ~Transport() {}
+};
+
+} // namespace webrtc
+
+#endif // API_CALL_TRANSPORT_H_