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- // Copyright 2019 The Chromium Authors. All rights reserved.
- // Use of this source code is governed by a BSD-style license that can be
- // found in the LICENSE file.
- #include "remoting/test/fake_webrtc_connection.h"
- #include "components/webrtc/thread_wrapper.h"
- #include "remoting/base/logging.h"
- #include "remoting/protocol/transport_context.h"
- namespace remoting {
- namespace test {
- FakeWebrtcConnection::FakeWebrtcConnection(
- scoped_refptr<protocol::TransportContext> transport_context,
- base::OnceClosure on_closed) {
- // TODO(lambroslambrou): Passing nullptr for the VideoEncoderFactory may
- // break the ftl_signaling_playground executable. If needed, this should be
- // replaced with a factory that supports at least one video codec.
- transport_ = std::make_unique<protocol::WebrtcTransport>(
- webrtc::ThreadWrapper::current(), transport_context, nullptr, this);
- on_closed_ = std::move(on_closed);
- }
- FakeWebrtcConnection::~FakeWebrtcConnection() = default;
- void FakeWebrtcConnection::OnWebrtcTransportConnecting() {
- HOST_LOG << "Webrtc transport is connecting...";
- }
- void FakeWebrtcConnection::OnWebrtcTransportConnected() {
- HOST_LOG << "Webrtc transport is connected!!!";
- std::move(on_closed_).Run();
- }
- void FakeWebrtcConnection::OnWebrtcTransportError(protocol::ErrorCode error) {
- LOG(ERROR) << "Webrtc transport error: " << error;
- std::move(on_closed_).Run();
- }
- void FakeWebrtcConnection::OnWebrtcTransportProtocolChanged() {}
- void FakeWebrtcConnection::OnWebrtcTransportIncomingDataChannel(
- const std::string& name,
- std::unique_ptr<protocol::MessagePipe> pipe) {}
- void FakeWebrtcConnection::OnWebrtcTransportMediaStreamAdded(
- rtc::scoped_refptr<webrtc::MediaStreamInterface> stream) {}
- void FakeWebrtcConnection::OnWebrtcTransportMediaStreamRemoved(
- rtc::scoped_refptr<webrtc::MediaStreamInterface> stream) {}
- void FakeWebrtcConnection::OnWebrtcTransportRouteChanged(
- const protocol::TransportRoute& route) {}
- } // namespace test
- } // namespace remoting
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