9c771c2089
This method will be called when PacedSender is using the new code path that directly owns the packets to be sent. It can be seen as combining a few features of the old code path: * It checks if this is the correct RTP module and then sends, without the need for PacketRouter to poll multiple methods for SSRC etc first. * It partly corresponds to TimeToSendPacket(), but RTX encapsulation now happens pre-pacer and FEC does not need to have a packet history, so most of that method is not used. * It implements most of PrepareAndSendPacket(), such as updating header extensions, reporting stats and of course forwards to Transport. It now also handles the history a bit differently, since media packets will only be stored for potential retransmission post-pacer. Bug: webrtc:10633 Change-Id: Ie97952eeef6e56e462e115d67f7c7929f36c1817 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142165 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28298}
112 lines
4.0 KiB
C++
112 lines
4.0 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 "absl/types/optional.h"
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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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enum class Type {
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kAudio, // Audio media packets.
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kVideo, // Video media packets.
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kRetransmission, // RTX (usually) packets send as response to NACK.
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kForwardErrorCorrection, // FEC packets.
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kPadding // RTX or plain padding sent to maintain BWE.
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};
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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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void set_packet_type(Type type) { packet_type_ = type; }
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absl::optional<Type> packet_type() const { return packet_type_; }
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// If this is a retransmission, indicates the sequence number of the original
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// media packet that this packet represents. If RTX is used this will likely
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// be different from SequenceNumber().
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void set_retransmitted_sequence_number(uint16_t sequence_number) {
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retransmitted_sequence_number_ = sequence_number;
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}
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absl::optional<uint16_t> retransmitted_sequence_number() {
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return retransmitted_sequence_number_;
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}
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void set_allow_retransmission(bool allow_retransmission) {
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allow_retransmission_ = allow_retransmission;
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}
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bool allow_retransmission() { return allow_retransmission_; }
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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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absl::optional<Type> packet_type_;
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bool allow_retransmission_ = false;
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absl::optional<uint16_t> retransmitted_sequence_number_;
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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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