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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
62 lines
2.2 KiB
C++
62 lines
2.2 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_TO_SEND_H_
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#define MODULES_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_
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#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
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#include "modules/rtp_rtcp/source/rtp_packet.h"
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namespace webrtc {
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// Class to hold rtp packet with metadata for sender side.
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class RtpPacketToSend : public RtpPacket {
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public:
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explicit RtpPacketToSend(const ExtensionManager* extensions)
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: RtpPacket(extensions) {}
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RtpPacketToSend(const RtpPacketToSend& packet) = default;
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RtpPacketToSend(const ExtensionManager* extensions, size_t capacity)
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: RtpPacket(extensions, capacity) {}
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RtpPacketToSend& operator=(const RtpPacketToSend& packet) = default;
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// Time in local time base as close as it can to frame capture time.
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int64_t capture_time_ms() const { return capture_time_ms_; }
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void set_capture_time_ms(int64_t time) { capture_time_ms_ = time; }
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void set_packetization_finish_time_ms(int64_t time) {
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SetExtension<VideoTimingExtension>(
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VideoSendTiming::GetDeltaCappedMs(capture_time_ms_, time),
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VideoSendTiming::kPacketizationFinishDeltaOffset);
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}
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void set_pacer_exit_time_ms(int64_t time) {
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SetExtension<VideoTimingExtension>(
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VideoSendTiming::GetDeltaCappedMs(capture_time_ms_, time),
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VideoSendTiming::kPacerExitDeltaOffset);
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}
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void set_network_time_ms(int64_t time) {
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SetExtension<VideoTimingExtension>(
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VideoSendTiming::GetDeltaCappedMs(capture_time_ms_, time),
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VideoSendTiming::kNetworkTimestampDeltaOffset);
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}
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void set_network2_time_ms(int64_t time) {
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SetExtension<VideoTimingExtension>(
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VideoSendTiming::GetDeltaCappedMs(capture_time_ms_, time),
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VideoSendTiming::kNetwork2TimestampDeltaOffset);
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}
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private:
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int64_t capture_time_ms_ = 0;
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};
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} // namespace webrtc
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#endif // MODULES_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_
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