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diff --git a/third_party/libwebrtc/call/call_factory.cc b/third_party/libwebrtc/call/call_factory.cc
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+++ b/third_party/libwebrtc/call/call_factory.cc
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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 <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();
+
+ 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<CallFactoryInterface> CreateCallFactory() {
+ return std::unique_ptr<CallFactoryInterface>(new CallFactory());
+}
+
+} // namespace webrtc