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119 lines
4 KiB
C++
119 lines
4 KiB
C++
/*
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* Copyright 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "call/call_factory.h"
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#include <stdio.h>
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#include <memory>
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#include <string>
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#include <utility>
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#include "absl/types/optional.h"
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#include "api/test/simulated_network.h"
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#include "call/call.h"
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#include "call/degraded_call.h"
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#include "call/rtp_transport_config.h"
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#include "rtc_base/checks.h"
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#include "system_wrappers/include/field_trial.h"
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namespace webrtc {
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namespace {
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bool ParseConfigParam(std::string exp_name, int* field) {
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std::string group = field_trial::FindFullName(exp_name);
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if (group.empty())
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return false;
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return (sscanf(group.c_str(), "%d", field) == 1);
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}
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absl::optional<webrtc::BuiltInNetworkBehaviorConfig> ParseDegradationConfig(
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bool send) {
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std::string exp_prefix = "WebRTCFakeNetwork";
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if (send) {
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exp_prefix += "Send";
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} else {
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exp_prefix += "Receive";
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}
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webrtc::BuiltInNetworkBehaviorConfig config;
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bool configured = false;
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configured |=
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ParseConfigParam(exp_prefix + "DelayMs", &config.queue_delay_ms);
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configured |= ParseConfigParam(exp_prefix + "DelayStdDevMs",
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&config.delay_standard_deviation_ms);
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int queue_length = 0;
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if (ParseConfigParam(exp_prefix + "QueueLength", &queue_length)) {
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RTC_CHECK_GE(queue_length, 0);
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config.queue_length_packets = queue_length;
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configured = true;
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}
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configured |=
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ParseConfigParam(exp_prefix + "CapacityKbps", &config.link_capacity_kbps);
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configured |=
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ParseConfigParam(exp_prefix + "LossPercent", &config.loss_percent);
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int allow_reordering = 0;
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if (ParseConfigParam(exp_prefix + "AllowReordering", &allow_reordering)) {
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config.allow_reordering = true;
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configured = true;
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}
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configured |= ParseConfigParam(exp_prefix + "AvgBurstLossLength",
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&config.avg_burst_loss_length);
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return configured
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? absl::optional<webrtc::BuiltInNetworkBehaviorConfig>(config)
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: absl::nullopt;
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}
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} // namespace
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CallFactory::CallFactory() {
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call_thread_.Detach();
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}
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Call* CallFactory::CreateCall(const Call::Config& config) {
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RTC_DCHECK_RUN_ON(&call_thread_);
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absl::optional<webrtc::BuiltInNetworkBehaviorConfig> send_degradation_config =
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ParseDegradationConfig(true);
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absl::optional<webrtc::BuiltInNetworkBehaviorConfig>
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receive_degradation_config = ParseDegradationConfig(false);
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RtpTransportConfig transportConfig = config.ExtractTransportConfig();
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if (send_degradation_config || receive_degradation_config) {
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return new DegradedCall(
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std::unique_ptr<Call>(Call::Create(
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config, Clock::GetRealTimeClock(),
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SharedModuleThread::Create(
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ProcessThread::Create("ModuleProcessThread"), nullptr),
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config.rtp_transport_controller_send_factory->Create(
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transportConfig, Clock::GetRealTimeClock(),
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ProcessThread::Create("PacerThread")))),
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send_degradation_config, receive_degradation_config,
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config.task_queue_factory);
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}
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if (!module_thread_) {
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module_thread_ = SharedModuleThread::Create(
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ProcessThread::Create("SharedModThread"), [this]() {
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RTC_DCHECK_RUN_ON(&call_thread_);
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module_thread_ = nullptr;
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});
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}
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return Call::Create(config, Clock::GetRealTimeClock(), module_thread_,
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config.rtp_transport_controller_send_factory->Create(
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transportConfig, Clock::GetRealTimeClock(),
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ProcessThread::Create("PacerThread")));
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}
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std::unique_ptr<CallFactoryInterface> CreateCallFactory() {
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return std::unique_ptr<CallFactoryInterface>(new CallFactory());
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}
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} // namespace webrtc
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