blob: b40aabdc2c6a0d3750f1c6956b8d22c12924e187 [file] [log] [blame]
* Copyright (c) 2017 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.
#include <stddef.h>
#include <stdint.h>
#include <map>
#include <memory>
#include <string>
#include <vector>
#include "absl/types/optional.h"
#include "api/crypto/crypto_options.h"
#include "api/fec_controller.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/transport/bitrate_settings.h"
#include "api/units/timestamp.h"
#include "call/rtp_config.h"
#include "modules/rtp_rtcp/include/report_block_data.h"
#include "modules/rtp_rtcp/include/rtcp_statistics.h"
#include "modules/rtp_rtcp/include/rtp_packet_sender.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
namespace rtc {
struct SentPacket;
struct NetworkRoute;
class TaskQueue;
} // namespace rtc
namespace webrtc {
class CallStatsObserver;
class FrameEncryptorInterface;
class TargetTransferRateObserver;
class Transport;
class Module;
class PacedSender;
class PacketRouter;
class RtpVideoSenderInterface;
class RateLimiter;
class RtcpBandwidthObserver;
class RtpPacketSender;
class SendDelayStats;
class SendStatisticsProxy;
struct RtpSenderObservers {
RtcpRttStats* rtcp_rtt_stats;
RtcpIntraFrameObserver* intra_frame_callback;
RtcpLossNotificationObserver* rtcp_loss_notification_observer;
RtcpStatisticsCallback* rtcp_stats;
ReportBlockDataObserver* report_block_data_observer;
StreamDataCountersCallback* rtp_stats;
BitrateStatisticsObserver* bitrate_observer;
FrameCountObserver* frame_count_observer;
RtcpPacketTypeCounterObserver* rtcp_type_observer;
SendSideDelayObserver* send_delay_observer;
SendPacketObserver* send_packet_observer;
struct RtpSenderFrameEncryptionConfig {
FrameEncryptorInterface* frame_encryptor = nullptr;
CryptoOptions crypto_options;
// An RtpTransportController should own everything related to the RTP
// transport to/from a remote endpoint. We should have separate
// interfaces for send and receive side, even if they are implemented
// by the same class. This is an ongoing refactoring project. At some
// point, this class should be promoted to a public api under
// webrtc/api/rtp/.
// For a start, this object is just a collection of the objects needed
// by the VideoSendStream constructor. The plan is to move ownership
// of all RTP-related objects here, and add methods to create per-ssrc
// objects which would then be passed to VideoSendStream. Eventually,
// direct accessors like packet_router() should be removed.
// This should also have a reference to the underlying
// webrtc::Transport(s). Currently, webrtc::Transport is implemented by
// WebRtcVideoChannel and WebRtcVoiceMediaChannel, and owned by
// WebrtcSession. Video and audio always uses different transport
// objects, even in the common case where they are bundled over the
// same underlying transport.
// Extracting the logic of the webrtc::Transport from BaseChannel and
// subclasses into a separate class seems to be a prerequesite for
// moving the transport here.
class RtpTransportControllerSendInterface {
virtual ~RtpTransportControllerSendInterface() {}
virtual rtc::TaskQueue* GetWorkerQueue() = 0;
virtual PacketRouter* packet_router() = 0;
virtual RtpVideoSenderInterface* CreateRtpVideoSender(
std::map<uint32_t, RtpState> suspended_ssrcs,
// TODO(holmer): Move states into RtpTransportControllerSend.
const std::map<uint32_t, RtpPayloadState>& states,
const RtpConfig& rtp_config,
int rtcp_report_interval_ms,
Transport* send_transport,
const RtpSenderObservers& observers,
RtcEventLog* event_log,
std::unique_ptr<FecController> fec_controller,
const RtpSenderFrameEncryptionConfig& frame_encryption_config) = 0;
virtual void DestroyRtpVideoSender(
RtpVideoSenderInterface* rtp_video_sender) = 0;
virtual NetworkStateEstimateObserver* network_state_estimate_observer() = 0;
virtual TransportFeedbackObserver* transport_feedback_observer() = 0;
virtual RtpPacketSender* packet_sender() = 0;
// SetAllocatedSendBitrateLimits sets bitrates limits imposed by send codec
// settings.
virtual void SetAllocatedSendBitrateLimits(
BitrateAllocationLimits limits) = 0;
virtual void SetPacingFactor(float pacing_factor) = 0;
virtual void SetQueueTimeLimit(int limit_ms) = 0;
virtual StreamFeedbackProvider* GetStreamFeedbackProvider() = 0;
virtual void RegisterTargetTransferRateObserver(
TargetTransferRateObserver* observer) = 0;
virtual void OnNetworkRouteChanged(
const std::string& transport_name,
const rtc::NetworkRoute& network_route) = 0;
virtual void OnNetworkAvailability(bool network_available) = 0;
virtual RtcpBandwidthObserver* GetBandwidthObserver() = 0;
virtual int64_t GetPacerQueuingDelayMs() const = 0;
virtual absl::optional<Timestamp> GetFirstPacketTime() const = 0;
virtual void EnablePeriodicAlrProbing(bool enable) = 0;
virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0;
virtual void OnReceivedPacket(const ReceivedPacket& received_packet) = 0;
virtual void SetSdpBitrateParameters(
const BitrateConstraints& constraints) = 0;
virtual void SetClientBitratePreferences(
const BitrateSettings& preferences) = 0;
virtual void OnTransportOverheadChanged(
size_t transport_overhead_per_packet) = 0;
virtual void AccountForAudioPacketsInPacedSender(bool account_for_audio) = 0;
virtual void IncludeOverheadInPacedSender() = 0;
} // namespace webrtc