blob: bc8a320f3d8a9d26ba4c20f35f683064f5f3a2b1 [file] [log] [blame]
/*
* Copyright 2018 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 API_TEST_FAKE_MEDIA_TRANSPORT_H_
#define API_TEST_FAKE_MEDIA_TRANSPORT_H_
#include <memory>
#include <string>
#include <utility>
#include <vector>
#include "absl/memory/memory.h"
#include "api/media_transport_interface.h"
namespace webrtc {
// TODO(sukhanov): For now fake media transport does nothing and is used only
// in jsepcontroller unittests. In the future we should implement fake media
// transport, which forwards frames to another fake media transport, so we
// could unit test audio / video integration.
class FakeMediaTransport : public MediaTransportInterface {
public:
explicit FakeMediaTransport(const MediaTransportSettings& settings)
: settings_(settings) {}
~FakeMediaTransport() = default;
RTCError SendAudioFrame(uint64_t channel_id,
MediaTransportEncodedAudioFrame frame) override {
return RTCError::OK();
}
RTCError SendVideoFrame(
uint64_t channel_id,
const MediaTransportEncodedVideoFrame& frame) override {
return RTCError::OK();
}
RTCError RequestKeyFrame(uint64_t channel_id) override {
return RTCError::OK();
};
void SetReceiveAudioSink(MediaTransportAudioSinkInterface* sink) override {}
void SetReceiveVideoSink(MediaTransportVideoSinkInterface* sink) override {}
// Returns true if fake media transport was created as a caller.
bool is_caller() const { return settings_.is_caller; }
absl::optional<std::string> pre_shared_key() const {
return settings_.pre_shared_key;
}
RTCError SendData(int channel_id,
const SendDataParams& params,
const rtc::CopyOnWriteBuffer& buffer) override {
return RTCError::OK();
}
RTCError CloseChannel(int channel_id) override { return RTCError::OK(); }
void SetDataSink(DataChannelSink* sink) override {}
void SetMediaTransportStateCallback(
MediaTransportStateCallback* callback) override {
state_callback_ = callback;
}
void SetState(webrtc::MediaTransportState state) {
if (state_callback_) {
state_callback_->OnStateChanged(state);
}
}
void AddTargetTransferRateObserver(
webrtc::TargetTransferRateObserver* observer) override {
RTC_CHECK(std::find(target_rate_observers_.begin(),
target_rate_observers_.end(),
observer) == target_rate_observers_.end());
target_rate_observers_.push_back(observer);
}
void RemoveTargetTransferRateObserver(
webrtc::TargetTransferRateObserver* observer) override {
auto it = std::find(target_rate_observers_.begin(),
target_rate_observers_.end(), observer);
if (it != target_rate_observers_.end()) {
target_rate_observers_.erase(it);
}
}
void SetAllocatedBitrateLimits(
const MediaTransportAllocatedBitrateLimits& limits) override {}
int target_rate_observers_size() { return target_rate_observers_.size(); }
private:
const MediaTransportSettings settings_;
MediaTransportStateCallback* state_callback_;
std::vector<webrtc::TargetTransferRateObserver*> target_rate_observers_;
};
// Fake media transport factory creates fake media transport.
class FakeMediaTransportFactory : public MediaTransportFactory {
public:
FakeMediaTransportFactory() = default;
~FakeMediaTransportFactory() = default;
RTCErrorOr<std::unique_ptr<MediaTransportInterface>> CreateMediaTransport(
rtc::PacketTransportInternal* packet_transport,
rtc::Thread* network_thread,
bool is_caller) override {
MediaTransportSettings settings;
settings.is_caller = is_caller;
return CreateMediaTransport(packet_transport, network_thread, settings);
}
RTCErrorOr<std::unique_ptr<MediaTransportInterface>> CreateMediaTransport(
rtc::PacketTransportInternal* packet_transport,
rtc::Thread* network_thread,
const MediaTransportSettings& settings) override {
std::unique_ptr<MediaTransportInterface> media_transport =
absl::make_unique<FakeMediaTransport>(settings);
return std::move(media_transport);
}
};
} // namespace webrtc
#endif // API_TEST_FAKE_MEDIA_TRANSPORT_H_