Add RtpPacketInfo to hold information about a received RtpPacket.
This change adds classes so that we later can plumb information about received packets to each audio and video frame. It's not wired up to do anything yet.
Bug: webrtc:10668
Change-Id: I962df493a76692f668314f78d6792d7636c5a31b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138203
Commit-Queue: Chen Xing <chxg@google.com>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28138}
diff --git a/api/rtp_packet_info.cc b/api/rtp_packet_info.cc
new file mode 100644
index 0000000..a61b173
--- /dev/null
+++ b/api/rtp_packet_info.cc
@@ -0,0 +1,58 @@
+/*
+ * 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), sequence_number_(0), rtp_timestamp_(0), receive_time_ms_(-1) {}
+
+RtpPacketInfo::RtpPacketInfo(uint32_t ssrc,
+ std::vector<uint32_t> csrcs,
+ uint16_t sequence_number,
+ uint32_t rtp_timestamp,
+ absl::optional<uint8_t> audio_level,
+ int64_t receive_time_ms)
+ : ssrc_(ssrc),
+ csrcs_(std::move(csrcs)),
+ sequence_number_(sequence_number),
+ rtp_timestamp_(rtp_timestamp),
+ audio_level_(audio_level),
+ receive_time_ms_(receive_time_ms) {}
+
+RtpPacketInfo::RtpPacketInfo(const RTPHeader& rtp_header,
+ int64_t receive_time_ms)
+ : ssrc_(rtp_header.ssrc),
+ sequence_number_(rtp_header.sequenceNumber),
+ 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;
+ }
+}
+
+bool operator==(const RtpPacketInfo& lhs, const RtpPacketInfo& rhs) {
+ return (lhs.ssrc() == rhs.ssrc()) && (lhs.csrcs() == rhs.csrcs()) &&
+ (lhs.sequence_number() == rhs.sequence_number()) &&
+ (lhs.rtp_timestamp() == rhs.rtp_timestamp()) &&
+ (lhs.audio_level() == rhs.audio_level()) &&
+ (lhs.receive_time_ms() == rhs.receive_time_ms());
+}
+
+} // namespace webrtc
