webrtc_m130/call/rtp_transport_controller_send.h
Sebastian Jansson c3eb9fd49f Reland "Reland "Only include overhead if using send side bandwidth estimation.""
This is a reland of 086055d0fd9b9b9efe8bcf85884324a019e9bd33

ANA was accitendly disabled even when transport sequence numbers were
negotiated due to a bug in how the audio send stream is configured. To
solve this we simply continue to always allow enabling ANA and leave it
up to the application to ensure that it's not used together with receive
side estimation.

Original change's description:
> Reland "Only include overhead if using send side bandwidth estimation."
>
> This is a reland of 8c79c6e1af354c526497082c79ccbe12af03a33e
>
> Original change's description:
> > Only include overhead if using send side bandwidth estimation.
> >
> > Bug: webrtc:11298
> > Change-Id: Ia2daf690461b55d394c1b964d6a7977a98be8be2
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166820
> > Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> > Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> > Reviewed-by: Ali Tofigh <alito@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30382}
>
> Bug: webrtc:11298
> Change-Id: I33205e869a8ae27c15ffe991f6d985973ed6d15a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167524
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30390}

Bug: webrtc:11298
Change-Id: If2ad91e17ebfc85dc51edcd9607996e18c5d1f13
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167883
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30413}
2020-01-29 18:42:34 +00:00

201 lines
8.1 KiB
C++

/*
* 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.
*/
#ifndef CALL_RTP_TRANSPORT_CONTROLLER_SEND_H_
#define CALL_RTP_TRANSPORT_CONTROLLER_SEND_H_
#include <atomic>
#include <map>
#include <memory>
#include <string>
#include <vector>
#include "api/network_state_predictor.h"
#include "api/transport/network_control.h"
#include "call/rtp_bitrate_configurator.h"
#include "call/rtp_transport_controller_send_interface.h"
#include "call/rtp_video_sender.h"
#include "modules/congestion_controller/rtp/control_handler.h"
#include "modules/congestion_controller/rtp/transport_feedback_adapter.h"
#include "modules/congestion_controller/rtp/transport_feedback_demuxer.h"
#include "modules/pacing/paced_sender.h"
#include "modules/pacing/packet_router.h"
#include "modules/pacing/rtp_packet_pacer.h"
#include "modules/pacing/task_queue_paced_sender.h"
#include "modules/utility/include/process_thread.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/network_route.h"
#include "rtc_base/race_checker.h"
#include "rtc_base/task_queue.h"
#include "rtc_base/task_utils/repeating_task.h"
namespace webrtc {
class Clock;
class FrameEncryptorInterface;
class RtcEventLog;
// TODO(nisse): When we get the underlying transports here, we should
// have one object implementing RtpTransportControllerSendInterface
// per transport, sharing the same congestion controller.
class RtpTransportControllerSend final
: public RtpTransportControllerSendInterface,
public RtcpBandwidthObserver,
public TransportFeedbackObserver,
public NetworkStateEstimateObserver {
public:
RtpTransportControllerSend(
Clock* clock,
RtcEventLog* event_log,
NetworkStatePredictorFactoryInterface* predictor_factory,
NetworkControllerFactoryInterface* controller_factory,
const BitrateConstraints& bitrate_config,
std::unique_ptr<ProcessThread> process_thread,
TaskQueueFactory* task_queue_factory,
const WebRtcKeyValueConfig* trials);
~RtpTransportControllerSend() override;
RtpVideoSenderInterface* CreateRtpVideoSender(
std::map<uint32_t, RtpState> suspended_ssrcs,
const std::map<uint32_t, RtpPayloadState>&
states, // move states into RtpTransportControllerSend
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) override;
void DestroyRtpVideoSender(
RtpVideoSenderInterface* rtp_video_sender) override;
// Implements RtpTransportControllerSendInterface
rtc::TaskQueue* GetWorkerQueue() override;
PacketRouter* packet_router() override;
NetworkStateEstimateObserver* network_state_estimate_observer() override;
TransportFeedbackObserver* transport_feedback_observer() override;
RtpPacketSender* packet_sender() override;
void SetAllocatedSendBitrateLimits(BitrateAllocationLimits limits) override;
void SetPacingFactor(float pacing_factor) override;
void SetQueueTimeLimit(int limit_ms) override;
StreamFeedbackProvider* GetStreamFeedbackProvider() override;
void RegisterTargetTransferRateObserver(
TargetTransferRateObserver* observer) override;
void OnNetworkRouteChanged(const std::string& transport_name,
const rtc::NetworkRoute& network_route) override;
void OnNetworkAvailability(bool network_available) override;
RtcpBandwidthObserver* GetBandwidthObserver() override;
int64_t GetPacerQueuingDelayMs() const override;
absl::optional<Timestamp> GetFirstPacketTime() const override;
void EnablePeriodicAlrProbing(bool enable) override;
void OnSentPacket(const rtc::SentPacket& sent_packet) override;
void OnReceivedPacket(const ReceivedPacket& packet_msg) override;
void SetSdpBitrateParameters(const BitrateConstraints& constraints) override;
void SetClientBitratePreferences(const BitrateSettings& preferences) override;
void OnTransportOverheadChanged(
size_t transport_overhead_per_packet) override;
void AccountForAudioPacketsInPacedSender(bool account_for_audio) override;
void IncludeOverheadInPacedSender() override;
// Implements RtcpBandwidthObserver interface
void OnReceivedEstimatedBitrate(uint32_t bitrate) override;
void OnReceivedRtcpReceiverReport(const ReportBlockList& report_blocks,
int64_t rtt,
int64_t now_ms) override;
// Implements TransportFeedbackObserver interface
void OnAddPacket(const RtpPacketSendInfo& packet_info) override;
void OnTransportFeedback(const rtcp::TransportFeedback& feedback) override;
// Implements NetworkStateEstimateObserver interface
void OnRemoteNetworkEstimate(NetworkStateEstimate estimate) override;
private:
void MaybeCreateControllers() RTC_RUN_ON(task_queue_);
void UpdateInitialConstraints(TargetRateConstraints new_contraints)
RTC_RUN_ON(task_queue_);
void StartProcessPeriodicTasks() RTC_RUN_ON(task_queue_);
void UpdateControllerWithTimeInterval() RTC_RUN_ON(task_queue_);
void UpdateStreamsConfig() RTC_RUN_ON(task_queue_);
void OnReceivedRtcpReceiverReportBlocks(const ReportBlockList& report_blocks,
int64_t now_ms)
RTC_RUN_ON(task_queue_);
void PostUpdates(NetworkControlUpdate update) RTC_RUN_ON(task_queue_);
void UpdateControlState() RTC_RUN_ON(task_queue_);
RtpPacketPacer* pacer();
const RtpPacketPacer* pacer() const;
Clock* const clock_;
RtcEventLog* const event_log_;
PacketRouter packet_router_;
std::vector<std::unique_ptr<RtpVideoSenderInterface>> video_rtp_senders_;
RtpBitrateConfigurator bitrate_configurator_;
std::map<std::string, rtc::NetworkRoute> network_routes_;
const std::unique_ptr<ProcessThread> process_thread_;
const bool use_task_queue_pacer_;
std::unique_ptr<PacedSender> process_thread_pacer_;
std::unique_ptr<TaskQueuePacedSender> task_queue_pacer_;
TargetTransferRateObserver* observer_ RTC_GUARDED_BY(task_queue_);
TransportFeedbackDemuxer feedback_demuxer_;
TransportFeedbackAdapter transport_feedback_adapter_
RTC_GUARDED_BY(task_queue_);
NetworkControllerFactoryInterface* const controller_factory_override_
RTC_PT_GUARDED_BY(task_queue_);
const std::unique_ptr<NetworkControllerFactoryInterface>
controller_factory_fallback_ RTC_PT_GUARDED_BY(task_queue_);
std::unique_ptr<CongestionControlHandler> control_handler_
RTC_GUARDED_BY(task_queue_) RTC_PT_GUARDED_BY(task_queue_);
std::unique_ptr<NetworkControllerInterface> controller_
RTC_GUARDED_BY(task_queue_) RTC_PT_GUARDED_BY(task_queue_);
TimeDelta process_interval_ RTC_GUARDED_BY(task_queue_);
std::map<uint32_t, RTCPReportBlock> last_report_blocks_
RTC_GUARDED_BY(task_queue_);
Timestamp last_report_block_time_ RTC_GUARDED_BY(task_queue_);
NetworkControllerConfig initial_config_ RTC_GUARDED_BY(task_queue_);
StreamsConfig streams_config_ RTC_GUARDED_BY(task_queue_);
const bool reset_feedback_on_route_change_;
const bool send_side_bwe_with_overhead_;
const bool add_pacing_to_cwin_;
size_t transport_overhead_bytes_per_packet_ RTC_GUARDED_BY(task_queue_);
bool network_available_ RTC_GUARDED_BY(task_queue_);
RepeatingTaskHandle pacer_queue_update_task_ RTC_GUARDED_BY(task_queue_);
RepeatingTaskHandle controller_task_ RTC_GUARDED_BY(task_queue_);
// Protected by internal locks.
RateLimiter retransmission_rate_limiter_;
// TODO(perkj): |task_queue_| is supposed to replace |process_thread_|.
// |task_queue_| is defined last to ensure all pending tasks are cancelled
// and deleted before any other members.
rtc::TaskQueue task_queue_;
RTC_DISALLOW_COPY_AND_ASSIGN(RtpTransportControllerSend);
};
} // namespace webrtc
#endif // CALL_RTP_TRANSPORT_CONTROLLER_SEND_H_