1 files changed, 98 insertions, 0 deletions
diff --git a/third_party/libwebrtc/call/create_call.cc b/third_party/libwebrtc/call/create_call.cc
new file mode 100644
index 0000000000..8b565745a8
--- /dev/null
+++ b/third_party/libwebrtc/call/create_call.cc
@@ -0,0 +1,98 @@
+/*
+ *  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/create_call.h"
+
+#include <stdio.h>
+
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#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 "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
+
+std::unique_ptr<Call> CreateCall(const CallConfig& config) {
+  std::vector<DegradedCall::TimeScopedNetworkConfig> send_degradation_configs =
+      GetNetworkConfigs(config.env.field_trials(), /*send=*/true);
+  std::vector<DegradedCall::TimeScopedNetworkConfig>
+      receive_degradation_configs =
+          GetNetworkConfigs(config.env.field_trials(), /*send=*/false);
+
+  std::unique_ptr<Call> call = Call::Create(config);
+
+  if (!send_degradation_configs.empty() ||
+      !receive_degradation_configs.empty()) {
+    return std::make_unique<DegradedCall>(
+        std::move(call), send_degradation_configs, receive_degradation_configs);
+  }
+
+  return call;
+}
+
+}  // namespace webrtc