blob: 380e80ce124028877e93c616692855473449814f [file] [log] [blame]
/*
* 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 <stdio.h>
#include <memory>
#include <string>
#include <utility>
#include <vector>
#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<TimeScopedNetworkConfig> GetNetworkConfigs(
const FieldTrialsView& trials,
bool send) {
FieldTrialStructList<TimeScopedNetworkConfig> 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<unsigned*>(
&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<DegradedCall::TimeScopedNetworkConfig> send_degradation_configs =
GetNetworkConfigs(*config.trials, /*send=*/true);
std::vector<DegradedCall::TimeScopedNetworkConfig>
receive_degradation_configs =
GetNetworkConfigs(*config.trials, /*send=*/false);
RtpTransportConfig transportConfig = config.ExtractTransportConfig();
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<CallFactoryInterface> CreateCallFactory() {
return std::unique_ptr<CallFactoryInterface>(new CallFactory());
}
} // namespace webrtc