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a4d873786f
This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505}
76 lines
2.6 KiB
C++
76 lines
2.6 KiB
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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#ifndef MODULES_RTP_RTCP_SOURCE_RTP_PACKET_RECEIVED_H_
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#define MODULES_RTP_RTCP_SOURCE_RTP_PACKET_RECEIVED_H_
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#include <stdint.h>
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#include <vector>
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#include "api/array_view.h"
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#include "api/rtp_headers.h"
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#include "modules/rtp_rtcp/source/rtp_packet.h"
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#include "system_wrappers/include/ntp_time.h"
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namespace webrtc {
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// Class to hold rtp packet with metadata for receiver side.
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class RtpPacketReceived : public RtpPacket {
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public:
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RtpPacketReceived();
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explicit RtpPacketReceived(const ExtensionManager* extensions);
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RtpPacketReceived(const RtpPacketReceived& packet);
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RtpPacketReceived(RtpPacketReceived&& packet);
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RtpPacketReceived& operator=(const RtpPacketReceived& packet);
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RtpPacketReceived& operator=(RtpPacketReceived&& packet);
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~RtpPacketReceived();
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// TODO(danilchap): Remove this function when all code update to use RtpPacket
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// directly. Function is there just for easier backward compatibilty.
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void GetHeader(RTPHeader* header) const;
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// Time in local time base as close as it can to packet arrived on the
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// network.
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int64_t arrival_time_ms() const { return arrival_time_ms_; }
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void set_arrival_time_ms(int64_t time) { arrival_time_ms_ = time; }
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// Estimated from Timestamp() using rtcp Sender Reports.
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NtpTime capture_ntp_time() const { return capture_time_; }
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void set_capture_ntp_time(NtpTime time) { capture_time_ = time; }
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// Flag if packet was recovered via RTX or FEC.
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bool recovered() const { return recovered_; }
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void set_recovered(bool value) { recovered_ = value; }
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int payload_type_frequency() const { return payload_type_frequency_; }
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void set_payload_type_frequency(int value) {
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payload_type_frequency_ = value;
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}
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// Additional data bound to the RTP packet for use in application code,
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// outside of WebRTC.
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rtc::ArrayView<const uint8_t> application_data() const {
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return application_data_;
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}
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void set_application_data(rtc::ArrayView<const uint8_t> data) {
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application_data_.assign(data.begin(), data.end());
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}
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private:
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NtpTime capture_time_;
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int64_t arrival_time_ms_ = 0;
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int payload_type_frequency_ = 0;
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bool recovered_ = false;
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std::vector<uint8_t> application_data_;
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};
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} // namespace webrtc
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#endif // MODULES_RTP_RTCP_SOURCE_RTP_PACKET_RECEIVED_H_
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