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/*
* 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 "absl/types/optional.h"
#include "api/test/simulated_network.h"
#include "call/call.h"
#include "call/degraded_call.h"
#include "call/rtp_transport_config.h"
#include "rtc_base/checks.h"
#include "system_wrappers/include/field_trial.h"
namespace webrtc {
namespace {
bool ParseConfigParam(std::string exp_name, int* field) {
std::string group = field_trial::FindFullName(exp_name);
if (group.empty())
return false;
return (sscanf(group.c_str(), "%d", field) == 1);
}
absl::optional<webrtc::BuiltInNetworkBehaviorConfig> ParseDegradationConfig(
bool send) {
std::string exp_prefix = "WebRTCFakeNetwork";
if (send) {
exp_prefix += "Send";
} else {
exp_prefix += "Receive";
}
webrtc::BuiltInNetworkBehaviorConfig config;
bool configured = false;
configured |=
ParseConfigParam(exp_prefix + "DelayMs", &config.queue_delay_ms);
configured |= ParseConfigParam(exp_prefix + "DelayStdDevMs",
&config.delay_standard_deviation_ms);
int queue_length = 0;
if (ParseConfigParam(exp_prefix + "QueueLength", &queue_length)) {
RTC_CHECK_GE(queue_length, 0);
config.queue_length_packets = queue_length;
configured = true;
}
configured |=
ParseConfigParam(exp_prefix + "CapacityKbps", &config.link_capacity_kbps);
configured |=
ParseConfigParam(exp_prefix + "LossPercent", &config.loss_percent);
int allow_reordering = 0;
if (ParseConfigParam(exp_prefix + "AllowReordering", &allow_reordering)) {
config.allow_reordering = true;
configured = true;
}
configured |= ParseConfigParam(exp_prefix + "AvgBurstLossLength",
&config.avg_burst_loss_length);
return configured
? absl::optional<webrtc::BuiltInNetworkBehaviorConfig>(config)
: absl::nullopt;
}
} // namespace
CallFactory::CallFactory() {
call_thread_.Detach();
}
Call* CallFactory::CreateCall(const Call::Config& config) {
RTC_DCHECK_RUN_ON(&call_thread_);
absl::optional<webrtc::BuiltInNetworkBehaviorConfig> send_degradation_config =
ParseDegradationConfig(true);
absl::optional<webrtc::BuiltInNetworkBehaviorConfig>
receive_degradation_config = ParseDegradationConfig(false);
RtpTransportConfig transportConfig = config.ExtractTransportConfig();
if (send_degradation_config || receive_degradation_config) {
return new DegradedCall(
std::unique_ptr<Call>(Call::Create(
config, Clock::GetRealTimeClock(),
SharedModuleThread::Create(
ProcessThread::Create("ModuleProcessThread"), nullptr),
config.rtp_transport_controller_send_factory->Create(
transportConfig, Clock::GetRealTimeClock(),
ProcessThread::Create("PacerThread")))),
send_degradation_config, receive_degradation_config,
config.task_queue_factory);
}
if (!module_thread_) {
module_thread_ = SharedModuleThread::Create(
ProcessThread::Create("SharedModThread"), [this]() {
RTC_DCHECK_RUN_ON(&call_thread_);
module_thread_ = nullptr;
});
}
return Call::Create(config, Clock::GetRealTimeClock(), module_thread_,
config.rtp_transport_controller_send_factory->Create(
transportConfig, Clock::GetRealTimeClock(),
ProcessThread::Create("PacerThread")));
}
std::unique_ptr<CallFactoryInterface> CreateCallFactory() {
return std::unique_ptr<CallFactoryInterface>(new CallFactory());
}
} // namespace webrtc
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