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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 "call/rtp_transport_controller_send_interface.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/congestion_controller/include/send_side_congestion_controller.h"
#include "rtc_base/constructormagic.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:
RtpTransportControllerSend(Clock* clock, webrtc::RtcEventLog* event_log);
// Implements RtpTransportControllerSendInterface
PacketRouter* packet_router() override;
// TODO(holmer): Temporarily exposed, should be removed and the
// appropriate methods should be added to this class instead.
// In addition the PacedSender should be driven by this class, either
// by owning the process thread, or later by using a task queue.
PacedSender* pacer() override;
SendSideCongestionController* send_side_cc() 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) override;
void SetKeepAliveConfig(const RtpKeepAliveConfig& config);
private:
PacketRouter packet_router_;
PacedSender pacer_;
SendSideCongestionController send_side_cc_;
RtpKeepAliveConfig keepalive_;
RTC_DISALLOW_COPY_AND_ASSIGN(RtpTransportControllerSend);
};
} // namespace webrtc
#endif // CALL_RTP_TRANSPORT_CONTROLLER_SEND_H_