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nissecae45d02017-04-24 05:53:20 -07001/*
2 * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020011#ifndef CALL_RTP_TRANSPORT_CONTROLLER_SEND_INTERFACE_H_
12#define CALL_RTP_TRANSPORT_CONTROLLER_SEND_INTERFACE_H_
Sebastian Janssone4be6da2018-02-15 16:51:41 +010013#include <stddef.h>
14#include <stdint.h>
nissecae45d02017-04-24 05:53:20 -070015
Stefan Holmerdbdb3a02018-07-17 16:03:46 +020016#include <map>
Stefan Holmer64be7fa2018-10-04 15:21:55 +020017#include <memory>
Sebastian Jansson97f61ea2018-02-21 13:01:55 +010018#include <string>
Stefan Holmerdbdb3a02018-07-17 16:03:46 +020019#include <vector>
Sebastian Jansson97f61ea2018-02-21 13:01:55 +010020
Danil Chapovalovb9b146c2018-06-15 12:28:07 +020021#include "absl/types/optional.h"
Steve Anton10542f22019-01-11 09:11:00 -080022#include "api/crypto/crypto_options.h"
Stefan Holmer64be7fa2018-10-04 15:21:55 +020023#include "api/fec_controller.h"
Marina Cioceae77912b2020-02-27 16:16:55 +010024#include "api/frame_transformer_interface.h"
Danil Chapovalov83bbe912019-08-07 12:24:53 +020025#include "api/rtc_event_log/rtc_event_log.h"
Niels Möller0c4f7be2018-05-07 14:01:37 +020026#include "api/transport/bitrate_settings.h"
Erik Språng425d6aa2019-07-29 16:38:27 +020027#include "api/units/timestamp.h"
Stefan Holmerdbdb3a02018-07-17 16:03:46 +020028#include "call/rtp_config.h"
Niels Möllera8327d42020-08-25 10:28:50 +020029#include "common_video/frame_counts.h"
Henrik Boström87e3f9d2019-05-27 10:44:24 +020030#include "modules/rtp_rtcp/include/report_block_data.h"
Niels Möller53382cb2018-11-27 14:05:08 +010031#include "modules/rtp_rtcp/include/rtcp_statistics.h"
Erik Språngaa59eca2019-07-24 14:52:55 +020032#include "modules/rtp_rtcp/include/rtp_packet_sender.h"
Stefan Holmerdbdb3a02018-07-17 16:03:46 +020033#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
Ying Wang8b279102019-05-27 17:19:08 +020034#include "modules/rtp_rtcp/source/rtp_packet_received.h"
Sebastian Jansson97f61ea2018-02-21 13:01:55 +010035
Sebastian Janssone4be6da2018-02-15 16:51:41 +010036namespace rtc {
37struct SentPacket;
38struct NetworkRoute;
Sebastian Janssone6256052018-05-04 14:08:15 +020039class TaskQueue;
Sebastian Janssone4be6da2018-02-15 16:51:41 +010040} // namespace rtc
nissecae45d02017-04-24 05:53:20 -070041namespace webrtc {
42
Benjamin Wright192eeec2018-10-17 17:27:25 -070043class FrameEncryptorInterface;
Sebastian Jansson19704ec2018-03-12 15:59:12 +010044class TargetTransferRateObserver;
Stefan Holmerdbdb3a02018-07-17 16:03:46 +020045class Transport;
nissecae45d02017-04-24 05:53:20 -070046class PacketRouter;
Stefan Holmer9416ef82018-07-19 10:34:38 +020047class RtpVideoSenderInterface;
Sebastian Janssone4be6da2018-02-15 16:51:41 +010048class RtcpBandwidthObserver;
nissecae45d02017-04-24 05:53:20 -070049class RtpPacketSender;
nissecae45d02017-04-24 05:53:20 -070050
Stefan Holmerdbdb3a02018-07-17 16:03:46 +020051struct RtpSenderObservers {
52 RtcpRttStats* rtcp_rtt_stats;
53 RtcpIntraFrameObserver* intra_frame_callback;
Elad Alon0a8562e2019-04-09 11:55:13 +020054 RtcpLossNotificationObserver* rtcp_loss_notification_observer;
Stefan Holmerdbdb3a02018-07-17 16:03:46 +020055 RtcpStatisticsCallback* rtcp_stats;
Henrik Boström87e3f9d2019-05-27 10:44:24 +020056 ReportBlockDataObserver* report_block_data_observer;
Stefan Holmerdbdb3a02018-07-17 16:03:46 +020057 StreamDataCountersCallback* rtp_stats;
58 BitrateStatisticsObserver* bitrate_observer;
59 FrameCountObserver* frame_count_observer;
60 RtcpPacketTypeCounterObserver* rtcp_type_observer;
61 SendSideDelayObserver* send_delay_observer;
62 SendPacketObserver* send_packet_observer;
Stefan Holmerdbdb3a02018-07-17 16:03:46 +020063};
64
Benjamin Wright192eeec2018-10-17 17:27:25 -070065struct RtpSenderFrameEncryptionConfig {
66 FrameEncryptorInterface* frame_encryptor = nullptr;
67 CryptoOptions crypto_options;
68};
69
nissecae45d02017-04-24 05:53:20 -070070// An RtpTransportController should own everything related to the RTP
71// transport to/from a remote endpoint. We should have separate
72// interfaces for send and receive side, even if they are implemented
73// by the same class. This is an ongoing refactoring project. At some
74// point, this class should be promoted to a public api under
75// webrtc/api/rtp/.
76//
77// For a start, this object is just a collection of the objects needed
78// by the VideoSendStream constructor. The plan is to move ownership
79// of all RTP-related objects here, and add methods to create per-ssrc
80// objects which would then be passed to VideoSendStream. Eventually,
81// direct accessors like packet_router() should be removed.
82//
83// This should also have a reference to the underlying
84// webrtc::Transport(s). Currently, webrtc::Transport is implemented by
eladalonf1841382017-06-12 01:16:46 -070085// WebRtcVideoChannel and WebRtcVoiceMediaChannel, and owned by
nissecae45d02017-04-24 05:53:20 -070086// WebrtcSession. Video and audio always uses different transport
87// objects, even in the common case where they are bundled over the
88// same underlying transport.
89//
90// Extracting the logic of the webrtc::Transport from BaseChannel and
91// subclasses into a separate class seems to be a prerequesite for
92// moving the transport here.
93class RtpTransportControllerSendInterface {
94 public:
95 virtual ~RtpTransportControllerSendInterface() {}
Sebastian Janssone6256052018-05-04 14:08:15 +020096 virtual rtc::TaskQueue* GetWorkerQueue() = 0;
nissecae45d02017-04-24 05:53:20 -070097 virtual PacketRouter* packet_router() = 0;
Stefan Holmerdbdb3a02018-07-17 16:03:46 +020098
Stefan Holmer9416ef82018-07-19 10:34:38 +020099 virtual RtpVideoSenderInterface* CreateRtpVideoSender(
Stefan Holmerdbdb3a02018-07-17 16:03:46 +0200100 std::map<uint32_t, RtpState> suspended_ssrcs,
101 // TODO(holmer): Move states into RtpTransportControllerSend.
102 const std::map<uint32_t, RtpPayloadState>& states,
103 const RtpConfig& rtp_config,
Jiawei Ou55718122018-11-09 13:17:39 -0800104 int rtcp_report_interval_ms,
Stefan Holmerdbdb3a02018-07-17 16:03:46 +0200105 Transport* send_transport,
106 const RtpSenderObservers& observers,
Stefan Holmer64be7fa2018-10-04 15:21:55 +0200107 RtcEventLog* event_log,
Benjamin Wright192eeec2018-10-17 17:27:25 -0700108 std::unique_ptr<FecController> fec_controller,
Marina Cioceae77912b2020-02-27 16:16:55 +0100109 const RtpSenderFrameEncryptionConfig& frame_encryption_config,
110 rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) = 0;
Stefan Holmer9416ef82018-07-19 10:34:38 +0200111 virtual void DestroyRtpVideoSender(
112 RtpVideoSenderInterface* rtp_video_sender) = 0;
Stefan Holmerdbdb3a02018-07-17 16:03:46 +0200113
Sebastian Janssone1795f42019-07-24 11:38:03 +0200114 virtual NetworkStateEstimateObserver* network_state_estimate_observer() = 0;
nissecae45d02017-04-24 05:53:20 -0700115 virtual TransportFeedbackObserver* transport_feedback_observer() = 0;
116
Erik Språngaa59eca2019-07-24 14:52:55 +0200117 virtual RtpPacketSender* packet_sender() = 0;
Stefan Holmer5c8942a2017-08-22 16:16:44 +0200118
119 // SetAllocatedSendBitrateLimits sets bitrates limits imposed by send codec
120 // settings.
Sebastian Jansson93b1ea22019-09-18 18:31:52 +0200121 virtual void SetAllocatedSendBitrateLimits(
122 BitrateAllocationLimits limits) = 0;
Sebastian Janssone4be6da2018-02-15 16:51:41 +0100123
Sebastian Jansson4c1ffb82018-02-15 16:51:58 +0100124 virtual void SetPacingFactor(float pacing_factor) = 0;
125 virtual void SetQueueTimeLimit(int limit_ms) = 0;
126
Sebastian Janssonf2988552019-10-29 17:18:51 +0100127 virtual StreamFeedbackProvider* GetStreamFeedbackProvider() = 0;
Sebastian Jansson19704ec2018-03-12 15:59:12 +0100128 virtual void RegisterTargetTransferRateObserver(
129 TargetTransferRateObserver* observer) = 0;
Sebastian Jansson97f61ea2018-02-21 13:01:55 +0100130 virtual void OnNetworkRouteChanged(
131 const std::string& transport_name,
132 const rtc::NetworkRoute& network_route) = 0;
Sebastian Janssone4be6da2018-02-15 16:51:41 +0100133 virtual void OnNetworkAvailability(bool network_available) = 0;
Sebastian Janssone4be6da2018-02-15 16:51:41 +0100134 virtual RtcpBandwidthObserver* GetBandwidthObserver() = 0;
Sebastian Janssone4be6da2018-02-15 16:51:41 +0100135 virtual int64_t GetPacerQueuingDelayMs() const = 0;
Erik Språng425d6aa2019-07-29 16:38:27 +0200136 virtual absl::optional<Timestamp> GetFirstPacketTime() const = 0;
Sebastian Janssone4be6da2018-02-15 16:51:41 +0100137 virtual void EnablePeriodicAlrProbing(bool enable) = 0;
138 virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0;
Sebastian Jansson607a6f12019-06-13 17:48:53 +0200139 virtual void OnReceivedPacket(const ReceivedPacket& received_packet) = 0;
Sebastian Jansson97f61ea2018-02-21 13:01:55 +0100140
141 virtual void SetSdpBitrateParameters(
142 const BitrateConstraints& constraints) = 0;
143 virtual void SetClientBitratePreferences(
Niels Möller0c4f7be2018-05-07 14:01:37 +0200144 const BitrateSettings& preferences) = 0;
Alex Narestbcf91802018-06-25 16:08:36 +0200145
Stefan Holmer64be7fa2018-10-04 15:21:55 +0200146 virtual void OnTransportOverheadChanged(
147 size_t transport_overhead_per_packet) = 0;
Erik Språngaa59eca2019-07-24 14:52:55 +0200148
149 virtual void AccountForAudioPacketsInPacedSender(bool account_for_audio) = 0;
Sebastian Janssonc3eb9fd2020-01-29 17:42:52 +0100150 virtual void IncludeOverheadInPacedSender() = 0;
Erik Språng7703f232020-09-14 11:03:13 +0200151
152 virtual void EnsureStarted() = 0;
nissecae45d02017-04-24 05:53:20 -0700153};
154
155} // namespace webrtc
156
Mirko Bonadei92ea95e2017-09-15 06:47:31 +0200157#endif // CALL_RTP_TRANSPORT_CONTROLLER_SEND_INTERFACE_H_