776866774f
With this CL, the decision if an RTP packet should be sent as ect(1) is made in RtpControllerSend depending on if RFC 8888 has been negotiated and if CCFB is received with ECN enabled. Since webrtc does not yet adapt to ECN feedback, packets are sent as ECT(1) until the first feedback is received. Change-Id: Iddf63849328afbe54a7c8f921f2e8db134aeff6a Bug: webrtc:42225697 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367388 Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43609}
54 lines
1.5 KiB
C++
54 lines
1.5 KiB
C++
/*
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* Copyright (c) 2013 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 API_CALL_TRANSPORT_H_
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#define API_CALL_TRANSPORT_H_
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#include <stdint.h>
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#include "api/array_view.h"
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namespace webrtc {
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// TODO(holmer): Look into unifying this with the PacketOptions in
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// asyncpacketsocket.h.
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struct PacketOptions {
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PacketOptions();
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PacketOptions(const PacketOptions&);
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~PacketOptions();
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// Negative ids are invalid and should be interpreted
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// as packet_id not being set.
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int64_t packet_id = -1;
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// Whether this is an audio or video packet, excluding retransmissions.
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bool is_media = true;
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bool included_in_feedback = false;
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bool included_in_allocation = false;
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bool send_as_ect1 = false;
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// Whether this packet can be part of a packet batch at lower levels.
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bool batchable = false;
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// Whether this packet is the last of a batch.
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bool last_packet_in_batch = false;
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};
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class Transport {
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public:
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virtual bool SendRtp(rtc::ArrayView<const uint8_t> packet,
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const PacketOptions& options) = 0;
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virtual bool SendRtcp(rtc::ArrayView<const uint8_t> packet) = 0;
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protected:
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virtual ~Transport() {}
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};
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} // namespace webrtc
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#endif // API_CALL_TRANSPORT_H_
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