|  | /* | 
|  | *  Copyright (c) 2016 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. | 
|  | */ | 
|  | #ifndef MODULES_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_ | 
|  | #define MODULES_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_ | 
|  |  | 
|  | #include "modules/rtp_rtcp/source/rtp_header_extensions.h" | 
|  | #include "modules/rtp_rtcp/source/rtp_packet.h" | 
|  |  | 
|  | namespace webrtc { | 
|  | // Class to hold rtp packet with metadata for sender side. | 
|  | class RtpPacketToSend : public RtpPacket { | 
|  | public: | 
|  | explicit RtpPacketToSend(const ExtensionManager* extensions) | 
|  | : RtpPacket(extensions) {} | 
|  | RtpPacketToSend(const RtpPacketToSend& packet) = default; | 
|  | RtpPacketToSend(const ExtensionManager* extensions, size_t capacity) | 
|  | : RtpPacket(extensions, capacity) {} | 
|  |  | 
|  | RtpPacketToSend& operator=(const RtpPacketToSend& packet) = default; | 
|  |  | 
|  | // Time in local time base as close as it can to frame capture time. | 
|  | int64_t capture_time_ms() const { return capture_time_ms_; } | 
|  |  | 
|  | void set_capture_time_ms(int64_t time) { capture_time_ms_ = time; } | 
|  |  | 
|  | void set_packetization_finish_time_ms(int64_t time) { | 
|  | SetExtension<VideoTimingExtension>( | 
|  | VideoSendTiming::GetDeltaCappedMs(capture_time_ms_, time), | 
|  | VideoSendTiming::kPacketizationFinishDeltaOffset); | 
|  | } | 
|  |  | 
|  | void set_pacer_exit_time_ms(int64_t time) { | 
|  | SetExtension<VideoTimingExtension>( | 
|  | VideoSendTiming::GetDeltaCappedMs(capture_time_ms_, time), | 
|  | VideoSendTiming::kPacerExitDeltaOffset); | 
|  | } | 
|  |  | 
|  | void set_network_time_ms(int64_t time) { | 
|  | SetExtension<VideoTimingExtension>( | 
|  | VideoSendTiming::GetDeltaCappedMs(capture_time_ms_, time), | 
|  | VideoSendTiming::kNetworkTimestampDeltaOffset); | 
|  | } | 
|  |  | 
|  | void set_network2_time_ms(int64_t time) { | 
|  | SetExtension<VideoTimingExtension>( | 
|  | VideoSendTiming::GetDeltaCappedMs(capture_time_ms_, time), | 
|  | VideoSendTiming::kNetwork2TimestampDeltaOffset); | 
|  | } | 
|  |  | 
|  | private: | 
|  | int64_t capture_time_ms_ = 0; | 
|  | }; | 
|  |  | 
|  | }  // namespace webrtc | 
|  | #endif  // MODULES_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_ |