chromium/third_party/webrtc/pc/data_channel_controller.cc

/*
 *  Copyright 2019 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 "pc/data_channel_controller.h"

#include <cstdint>
#include <utility>

#include "absl/algorithm/container.h"
#include "absl/types/optional.h"
#include "api/peer_connection_interface.h"
#include "api/priority.h"
#include "api/rtc_error.h"
#include "pc/peer_connection_internal.h"
#include "pc/sctp_utils.h"
#include "rtc_base/logging.h"

namespace webrtc {

DataChannelController::~DataChannelController() {}

bool DataChannelController::HasDataChannels() const {}

bool DataChannelController::HasUsedDataChannels() const {}

RTCError DataChannelController::SendData(
    StreamId sid,
    const SendDataParams& params,
    const rtc::CopyOnWriteBuffer& payload) {}

void DataChannelController::AddSctpDataStream(StreamId sid,
                                              PriorityValue priority) {}

void DataChannelController::RemoveSctpDataStream(StreamId sid) {}

void DataChannelController::OnChannelStateChanged(
    SctpDataChannel* channel,
    DataChannelInterface::DataState state) {}

size_t DataChannelController::buffered_amount(StreamId sid) const {}

size_t DataChannelController::buffered_amount_low_threshold(
    StreamId sid) const {}

void DataChannelController::SetBufferedAmountLowThreshold(StreamId sid,
                                                          size_t bytes) {}

void DataChannelController::OnDataReceived(
    int channel_id,
    DataMessageType type,
    const rtc::CopyOnWriteBuffer& buffer) {}

void DataChannelController::OnChannelClosing(int channel_id) {}

void DataChannelController::OnChannelClosed(int channel_id) {}

void DataChannelController::OnReadyToSend() {}

void DataChannelController::OnTransportClosed(RTCError error) {}

void DataChannelController::OnBufferedAmountLow(int channel_id) {}

void DataChannelController::SetupDataChannelTransport_n(
    DataChannelTransportInterface* transport) {}

void DataChannelController::PrepareForShutdown() {}

void DataChannelController::TeardownDataChannelTransport_n(RTCError error) {}

void DataChannelController::OnTransportChanged(
    DataChannelTransportInterface* new_data_channel_transport) {}

std::vector<DataChannelStats> DataChannelController::GetDataChannelStats()
    const {}

bool DataChannelController::HandleOpenMessage_n(
    int channel_id,
    DataMessageType type,
    const rtc::CopyOnWriteBuffer& buffer) {}

void DataChannelController::OnDataChannelOpenMessage(
    rtc::scoped_refptr<SctpDataChannel> channel,
    bool ready_to_send) {}

// RTC_RUN_ON(network_thread())
RTCError DataChannelController::ReserveOrAllocateSid(
    absl::optional<StreamId>& sid,
    absl::optional<rtc::SSLRole> fallback_ssl_role) {}

// RTC_RUN_ON(network_thread())
RTCErrorOr<rtc::scoped_refptr<SctpDataChannel>>
DataChannelController::CreateDataChannel(const std::string& label,
                                         InternalDataChannelInit& config) {}

RTCErrorOr<rtc::scoped_refptr<DataChannelInterface>>
DataChannelController::InternalCreateDataChannelWithProxy(
    const std::string& label,
    const InternalDataChannelInit& config) {}

void DataChannelController::AllocateSctpSids(rtc::SSLRole role) {}

void DataChannelController::OnSctpDataChannelClosed(SctpDataChannel* channel) {}

void DataChannelController::set_data_channel_transport(
    DataChannelTransportInterface* transport) {}

void DataChannelController::NotifyDataChannelsOfTransportCreated() {}

rtc::Thread* DataChannelController::network_thread() const {}

rtc::Thread* DataChannelController::signaling_thread() const {}

}  // namespace webrtc