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https://github.com/mollyim/webrtc.git
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e08648dc70
This change stores the optional `AbsoluteCaptureTime` header extension in `RtpPacketInfo` so that we later can consume it in `SourceTracker`. Bug: webrtc:10739 Change-Id: I975e8863117fcda134535cd49ad71079a7ff38ec Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148068 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Chen Xing <chxg@google.com> Cr-Commit-Position: refs/heads/master@{#28790}
72 lines
2.5 KiB
C++
72 lines
2.5 KiB
C++
/*
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* Copyright (c) 2019 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 "api/rtp_packet_info.h"
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#include <algorithm>
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#include <utility>
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namespace webrtc {
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RtpPacketInfo::RtpPacketInfo()
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: ssrc_(0), rtp_timestamp_(0), receive_time_ms_(-1) {}
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RtpPacketInfo::RtpPacketInfo(
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uint32_t ssrc,
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std::vector<uint32_t> csrcs,
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uint32_t rtp_timestamp,
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absl::optional<uint8_t> audio_level,
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absl::optional<AbsoluteCaptureTime> absolute_capture_time,
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int64_t receive_time_ms)
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: ssrc_(ssrc),
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csrcs_(std::move(csrcs)),
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rtp_timestamp_(rtp_timestamp),
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audio_level_(audio_level),
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absolute_capture_time_(absolute_capture_time),
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receive_time_ms_(receive_time_ms) {}
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RtpPacketInfo::RtpPacketInfo(uint32_t ssrc,
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std::vector<uint32_t> csrcs,
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uint32_t rtp_timestamp,
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absl::optional<uint8_t> audio_level,
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int64_t receive_time_ms)
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: RtpPacketInfo(ssrc,
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std::move(csrcs),
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rtp_timestamp,
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audio_level,
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/*absolute_capture_time=*/absl::nullopt,
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receive_time_ms) {}
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RtpPacketInfo::RtpPacketInfo(const RTPHeader& rtp_header,
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int64_t receive_time_ms)
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: ssrc_(rtp_header.ssrc),
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rtp_timestamp_(rtp_header.timestamp),
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receive_time_ms_(receive_time_ms) {
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const auto& extension = rtp_header.extension;
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const auto csrcs_count = std::min<size_t>(rtp_header.numCSRCs, kRtpCsrcSize);
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csrcs_.assign(&rtp_header.arrOfCSRCs[0], &rtp_header.arrOfCSRCs[csrcs_count]);
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if (extension.hasAudioLevel) {
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audio_level_ = extension.audioLevel;
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}
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absolute_capture_time_ = extension.absolute_capture_time;
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}
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bool operator==(const RtpPacketInfo& lhs, const RtpPacketInfo& rhs) {
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return (lhs.ssrc() == rhs.ssrc()) && (lhs.csrcs() == rhs.csrcs()) &&
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(lhs.rtp_timestamp() == rhs.rtp_timestamp()) &&
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(lhs.audio_level() == rhs.audio_level()) &&
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(lhs.absolute_capture_time() == rhs.absolute_capture_time()) &&
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(lhs.receive_time_ms() == rhs.receive_time_ms());
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}
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} // namespace webrtc
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