/* * Copyright 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/call_factory.h" #include #include #include #include #include #include "absl/memory/memory.h" #include "absl/types/optional.h" #include "api/test/simulated_network.h" #include "api/units/time_delta.h" #include "call/call.h" #include "call/degraded_call.h" #include "call/rtp_transport_config.h" #include "rtc_base/checks.h" #include "rtc_base/experiments/field_trial_list.h" #include "rtc_base/experiments/field_trial_parser.h" namespace webrtc { namespace { using TimeScopedNetworkConfig = DegradedCall::TimeScopedNetworkConfig; std::vector GetNetworkConfigs( const FieldTrialsView& trials, bool send) { FieldTrialStructList trials_list( {FieldTrialStructMember("queue_length_packets", [](TimeScopedNetworkConfig* p) { // FieldTrialParser does not natively support // size_t type, so use this ugly cast as // workaround. return reinterpret_cast( &p->queue_length_packets); }), FieldTrialStructMember( "queue_delay_ms", [](TimeScopedNetworkConfig* p) { return &p->queue_delay_ms; }), FieldTrialStructMember("delay_standard_deviation_ms", [](TimeScopedNetworkConfig* p) { return &p->delay_standard_deviation_ms; }), FieldTrialStructMember( "link_capacity_kbps", [](TimeScopedNetworkConfig* p) { return &p->link_capacity_kbps; }), FieldTrialStructMember( "loss_percent", [](TimeScopedNetworkConfig* p) { return &p->loss_percent; }), FieldTrialStructMember( "allow_reordering", [](TimeScopedNetworkConfig* p) { return &p->allow_reordering; }), FieldTrialStructMember("avg_burst_loss_length", [](TimeScopedNetworkConfig* p) { return &p->avg_burst_loss_length; }), FieldTrialStructMember( "packet_overhead", [](TimeScopedNetworkConfig* p) { return &p->packet_overhead; }), FieldTrialStructMember( "duration", [](TimeScopedNetworkConfig* p) { return &p->duration; })}, {}); ParseFieldTrial({&trials_list}, trials.Lookup(send ? "WebRTC-FakeNetworkSendConfig" : "WebRTC-FakeNetworkReceiveConfig")); return trials_list.Get(); } } // namespace CallFactory::CallFactory() { call_thread_.Detach(); } Call* CallFactory::CreateCall(const Call::Config& config) { RTC_DCHECK_RUN_ON(&call_thread_); RTC_DCHECK(config.trials); std::vector send_degradation_configs = GetNetworkConfigs(*config.trials, /*send=*/true); std::vector receive_degradation_configs = GetNetworkConfigs(*config.trials, /*send=*/false); RtpTransportConfig transportConfig = config.ExtractTransportConfig(); RTC_CHECK(false); return nullptr; /* Mozilla: Avoid this since it could use GetRealTimeClock(). Call* call = Call::Create(config, Clock::GetRealTimeClock(), config.rtp_transport_controller_send_factory->Create( transportConfig, Clock::GetRealTimeClock())); if (!send_degradation_configs.empty() || !receive_degradation_configs.empty()) { return new DegradedCall(absl::WrapUnique(call), send_degradation_configs, receive_degradation_configs); } return call; */ } std::unique_ptr CreateCallFactory() { return std::unique_ptr(new CallFactory()); } } // namespace webrtc