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3e8ef940fe
This change adds the plumbing of RtpPacketInfo from ChannelReceive::OnRtpPacket() to ChannelReceive::GetAudioFrameWithInfo() for audio. It is a step towards replacing the non-spec compliant ContributingSources that updates itself at packet-receive time, with the spec-compliant SourceTracker that will update itself at frame-delivery-to-track time. Bug: webrtc:10668 Change-Id: I03385d6865bbc7bfbef7634f88de820a934f787a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139890 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Commit-Queue: Chen Xing <chxg@google.com> Cr-Commit-Position: refs/heads/master@{#28434}
36 lines
973 B
C++
36 lines
973 B
C++
/*
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* Copyright (c) 2016 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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#include "modules/audio_coding/neteq/packet.h"
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namespace webrtc {
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Packet::Packet() = default;
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Packet::Packet(Packet&& b) = default;
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Packet::~Packet() = default;
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Packet& Packet::operator=(Packet&& b) = default;
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Packet Packet::Clone() const {
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RTC_CHECK(!frame);
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Packet clone;
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clone.timestamp = timestamp;
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clone.sequence_number = sequence_number;
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clone.payload_type = payload_type;
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clone.payload.SetData(payload.data(), payload.size());
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clone.priority = priority;
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clone.packet_info = packet_info;
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return clone;
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}
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} // namespace webrtc
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