
This is a followup to https://webrtc-review.googlesource.com/c/src/+/91840, which needed transitional methods while updating downstream code. This cl completes the deletion, and can be landed after downstream code is updated. Bug: webtrc:9584 Change-Id: I4d3654748973a4757a8d79bb93f524c630a0eca3 Reviewed-on: https://webrtc-review.googlesource.com/93285 Commit-Queue: Oleh Prypin <oprypin@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24329}
43 lines
1.1 KiB
C++
43 lines
1.1 KiB
C++
/*
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* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef CALL_PACKET_RECEIVER_H_
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#define CALL_PACKET_RECEIVER_H_
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#include <algorithm>
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#include <memory>
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#include <string>
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#include <vector>
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#include "api/mediatypes.h"
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#include "common_types.h" // NOLINT(build/include)
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#include "rtc_base/copyonwritebuffer.h"
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namespace webrtc {
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class PacketReceiver {
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public:
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enum DeliveryStatus {
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DELIVERY_OK,
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DELIVERY_UNKNOWN_SSRC,
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DELIVERY_PACKET_ERROR,
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};
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virtual DeliveryStatus DeliverPacket(MediaType media_type,
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rtc::CopyOnWriteBuffer packet,
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int64_t packet_time_us) = 0;
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protected:
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virtual ~PacketReceiver() {}
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};
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} // namespace webrtc
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#endif // CALL_PACKET_RECEIVER_H_
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