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/*
* Copyright (c) 2019 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.
*/
#include "api/rtp_packet_info.h"
#include <algorithm>
#include <utility>
namespace webrtc {
RtpPacketInfo::RtpPacketInfo()
: ssrc_(0), rtp_timestamp_(0), receive_time_ms_(-1) {}
RtpPacketInfo::RtpPacketInfo(
uint32_t ssrc,
std::vector<uint32_t> csrcs,
uint32_t rtp_timestamp,
absl::optional<uint8_t> audio_level,
absl::optional<AbsoluteCaptureTime> absolute_capture_time,
int64_t receive_time_ms)
: ssrc_(ssrc),
csrcs_(std::move(csrcs)),
rtp_timestamp_(rtp_timestamp),
audio_level_(audio_level),
absolute_capture_time_(absolute_capture_time),
receive_time_ms_(receive_time_ms) {}
RtpPacketInfo::RtpPacketInfo(const RTPHeader& rtp_header,
int64_t receive_time_ms)
: ssrc_(rtp_header.ssrc),
rtp_timestamp_(rtp_header.timestamp),
receive_time_ms_(receive_time_ms) {
const auto& extension = rtp_header.extension;
const auto csrcs_count = std::min<size_t>(rtp_header.numCSRCs, kRtpCsrcSize);
csrcs_.assign(&rtp_header.arrOfCSRCs[0], &rtp_header.arrOfCSRCs[csrcs_count]);
if (extension.hasAudioLevel) {
audio_level_ = extension.audioLevel;
}
absolute_capture_time_ = extension.absolute_capture_time;
}
bool operator==(const RtpPacketInfo& lhs, const RtpPacketInfo& rhs) {
return (lhs.ssrc() == rhs.ssrc()) && (lhs.csrcs() == rhs.csrcs()) &&
(lhs.rtp_timestamp() == rhs.rtp_timestamp()) &&
(lhs.audio_level() == rhs.audio_level()) &&
(lhs.absolute_capture_time() == rhs.absolute_capture_time()) &&
(lhs.receive_time_ms() == rhs.receive_time_ms());
}
} // namespace webrtc
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