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https://github.com/mollyim/webrtc.git
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435ea0a741
Use instead of checking the packet's payload type and ssrc. Bug: webrtc:7135 Change-Id: I272922a7879ef3e5e1344ce49044688572b9d942 Reviewed-on: https://webrtc-review.googlesource.com/c/120048 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26425}
87 lines
2.9 KiB
C++
87 lines
2.9 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 <stddef.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/video/video_timing.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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RtpPacketToSend(const ExtensionManager* extensions, size_t capacity);
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RtpPacketToSend(const RtpPacketToSend& packet);
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RtpPacketToSend(RtpPacketToSend&& packet);
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RtpPacketToSend& operator=(const RtpPacketToSend& packet);
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RtpPacketToSend& operator=(RtpPacketToSend&& packet);
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~RtpPacketToSend();
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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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bool is_fec() const { return is_fec_; }
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void set_is_fec(bool fec) { is_fec_ = fec; }
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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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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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// Used for accounting purposes
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bool is_fec_ = 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_TO_SEND_H_
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