chromium/third_party/webrtc/call/rtp_stream_receiver_controller.cc

/*
 *  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 "call/rtp_stream_receiver_controller.h"

#include <memory>

#include "rtc_base/logging.h"

namespace webrtc {

RtpStreamReceiverController::Receiver::Receiver(
    RtpStreamReceiverController* controller,
    uint32_t ssrc,
    RtpPacketSinkInterface* sink)
    : … { … }

RtpStreamReceiverController::Receiver::~Receiver() { … }

RtpStreamReceiverController::RtpStreamReceiverController() { … }

RtpStreamReceiverController::~RtpStreamReceiverController() = default;

std::unique_ptr<RtpStreamReceiverInterface>
RtpStreamReceiverController::CreateReceiver(uint32_t ssrc,
                                            RtpPacketSinkInterface* sink) { … }

bool RtpStreamReceiverController::OnRtpPacket(const RtpPacketReceived& packet) { … }

void RtpStreamReceiverController::OnRecoveredPacket(
    const RtpPacketReceived& packet) { … }

bool RtpStreamReceiverController::AddSink(uint32_t ssrc,
                                          RtpPacketSinkInterface* sink) { … }

bool RtpStreamReceiverController::RemoveSink(
    const RtpPacketSinkInterface* sink) { … }

}  // namespace webrtc