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/*
* 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 <map>
#include <memory>
#include <string>
#include "call/rtp_bitrate_configurator.h"
#include "call/rtp_transport_controller_send_interface.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/congestion_controller/include/send_side_congestion_controller_interface.h"
#include "modules/pacing/packet_router.h"
#include "modules/utility/include/process_thread.h"
#include "rtc_base/constructormagic.h"
#include "rtc_base/networkroute.h"
namespace webrtc {
class Clock;
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 : public RtpTransportControllerSendInterface,
public NetworkChangedObserver {
public:
RtpTransportControllerSend(Clock* clock,
RtcEventLog* event_log,
const BitrateConstraints& bitrate_config);
~RtpTransportControllerSend() override;
// Implements NetworkChangedObserver interface.
void OnNetworkChanged(uint32_t bitrate_bps,
uint8_t fraction_loss,
int64_t rtt_ms,
int64_t probing_interval_ms) override;
// Implements RtpTransportControllerSendInterface
PacketRouter* packet_router() override;
TransportFeedbackObserver* transport_feedback_observer() override;
RtpPacketSender* packet_sender() override;
const RtpKeepAliveConfig& keepalive_config() const override;
void SetAllocatedSendBitrateLimits(int min_send_bitrate_bps,
int max_padding_bitrate_bps,
int max_total_bitrate_bps) override;
void SetKeepAliveConfig(const RtpKeepAliveConfig& config);
void SetPacingFactor(float pacing_factor) override;
void SetQueueTimeLimit(int limit_ms) override;
CallStatsObserver* GetCallStatsObserver() override;
void RegisterPacketFeedbackObserver(
PacketFeedbackObserver* observer) override;
void DeRegisterPacketFeedbackObserver(
PacketFeedbackObserver* observer) 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;
int64_t GetFirstPacketTimeMs() const override;
void EnablePeriodicAlrProbing(bool enable) override;
void OnSentPacket(const rtc::SentPacket& sent_packet) override;
void SetSdpBitrateParameters(const BitrateConstraints& constraints) override;
void SetClientBitratePreferences(
const BitrateConstraintsMask& preferences) override;
private:
const Clock* const clock_;
PacketRouter packet_router_;
PacedSender pacer_;
const std::unique_ptr<SendSideCongestionControllerInterface> send_side_cc_;
RtpKeepAliveConfig keepalive_;
RtpBitrateConfigurator bitrate_configurator_;
std::map<std::string, rtc::NetworkRoute> network_routes_;
const std::unique_ptr<ProcessThread> process_thread_;
rtc::CriticalSection observer_crit_;
TargetTransferRateObserver* observer_ RTC_GUARDED_BY(observer_crit_);
RTC_DISALLOW_COPY_AND_ASSIGN(RtpTransportControllerSend);
};
} // namespace webrtc
#endif // CALL_RTP_TRANSPORT_CONTROLLER_SEND_H_