Move call/simulated_network to test/network

Old target and call/simulated.h exist but refer to new target in test/network.

Bug: webrtc:14525
Change-Id: Ida04cef17913f2f829d7e925ae454dc40d5e8240
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349264
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Owners-Override: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42191}

api/BUILD.gn

@@ -865,7 +865,7 @@ rtc_source_set("network_emulation_manager_api") {
     ":peer_network_dependencies",
     ":simulated_network_api",
     ":time_controller",
-    "../call:simulated_network",
+    "..//test/network:simulated_network",
     "../rtc_base:checks",
     "../rtc_base:network",
     "../rtc_base:network_constants",

api/test/network_emulation_manager.cc

@@ -11,8 +11,8 @@
 
 #include <utility>
 
-#include "call/simulated_network.h"
 #include "rtc_base/checks.h"
+#include "test/network/simulated_network.h"
 
 namespace webrtc {
 

audio/BUILD.gn

@@ -134,7 +134,6 @@ if (rtc_include_tests) {
       "../api/audio:audio_device",
       "../api/task_queue",
       "../call:fake_network",
-      "../call:simulated_network",
       "../modules/audio_device:test_audio_device_module",
       "../system_wrappers",
       "../test:test_common",

audio/test/audio_end_to_end_test.cc

@@ -15,7 +15,6 @@
 
 #include "api/task_queue/task_queue_base.h"
 #include "call/fake_network_pipe.h"
-#include "call/simulated_network.h"
 #include "modules/audio_device/include/test_audio_device.h"
 #include "system_wrappers/include/sleep.h"
 #include "test/gtest.h"

call/BUILD.gn

@@ -336,6 +336,7 @@ rtc_library("call") {
     "../system_wrappers",
     "../system_wrappers:field_trial",
     "../system_wrappers:metrics",
+    "../test/network:simulated_network",
     "../video",
     "../video:decode_synchronizer",
     "../video/config:encoder_config",
@@ -399,24 +400,9 @@ rtc_library("video_stream_api") {
 }
 
 rtc_library("simulated_network") {
-  sources = [
+  # TODO(bugs.webrtc.org/14525): Remove target and directly depend on test/network:simulated_network.
-    "simulated_network.cc",
+  sources = [ "simulated_network.h" ]
-    "simulated_network.h",
+  deps = [ "../test/network:simulated_network" ]
-  ]
-  deps = [
-    "../api:sequence_checker",
-    "../api:simulated_network_api",
-    "../api/units:data_rate",
-    "../api/units:data_size",
-    "../api/units:time_delta",
-    "../api/units:timestamp",
-    "../rtc_base:checks",
-    "../rtc_base:macromagic",
-    "../rtc_base:race_checker",
-    "../rtc_base:random",
-    "../rtc_base/synchronization:mutex",
-  ]
-  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
 }
 
 rtc_source_set("simulated_packet_receiver") {
@@ -434,7 +420,6 @@ rtc_library("fake_network") {
   ]
   deps = [
     ":call_interfaces",
-    ":simulated_network",
     ":simulated_packet_receiver",
     "../api:rtp_parameters",
     "../api:sequence_checker",
@@ -447,6 +432,7 @@ rtc_library("fake_network") {
     "../rtc_base:macromagic",
     "../rtc_base/synchronization:mutex",
     "../system_wrappers",
+    "../test/network:simulated_network",
   ]
 }
 
@@ -606,6 +592,7 @@ if (rtc_include_tests) {
         "../test:test_support",
         "../test:video_test_common",
         "../test:video_test_constants",
+        "../test/network:simulated_network",
         "../video",
         "../video/config:encoder_config",
         "//testing/gtest",
@@ -663,13 +650,9 @@ if (rtc_include_tests) {
   rtc_library("fake_network_pipe_unittests") {
     testonly = true
 
-    sources = [
+    sources = [ "fake_network_pipe_unittest.cc" ]
-      "fake_network_pipe_unittest.cc",
-      "simulated_network_unittest.cc",
-    ]
     deps = [
       ":fake_network",
-      ":simulated_network",
       "../api:simulated_network_api",
       "../api/units:data_rate",
       "../api/units:time_delta",
@@ -678,6 +661,7 @@ if (rtc_include_tests) {
       "../rtc_base:checks",
       "../system_wrappers",
       "../test:test_support",
+      "../test/network:simulated_network",
       "//testing/gtest",
     ]
     absl_deps = [ "//third_party/abseil-cpp/absl/algorithm:container" ]

call/DEPS

@@ -28,5 +28,9 @@ specific_include_rules = {
   ],
   "call_perf_tests\.cc": [
     "+media/engine",
+  ],
+  "simulated_network\.h": [
+    "+test/network/simulated_network.h",
   ]
+
 }

call/bitrate_estimator_tests.cc

@@ -16,7 +16,6 @@
 #include "absl/strings/string_view.h"
 #include "api/test/create_frame_generator.h"
 #include "call/call.h"
-#include "call/simulated_network.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/event.h"
 #include "rtc_base/logging.h"

call/call_perf_tests.cc

@@ -29,7 +29,6 @@
 #include "api/video_codecs/video_encoder.h"
 #include "call/call.h"
 #include "call/fake_network_pipe.h"
-#include "call/simulated_network.h"
 #include "media/engine/internal_encoder_factory.h"
 #include "media/engine/simulcast_encoder_adapter.h"
 #include "modules/audio_coding/include/audio_coding_module.h"
@@ -51,6 +50,7 @@
 #include "test/field_trial.h"
 #include "test/frame_generator_capturer.h"
 #include "test/gtest.h"
+#include "test/network/simulated_network.h"
 #include "test/null_transport.h"
 #include "test/rtp_rtcp_observer.h"
 #include "test/test_flags.h"

call/degraded_call.h

@@ -34,12 +34,12 @@
 #include "call/flexfec_receive_stream.h"
 #include "call/packet_receiver.h"
 #include "call/rtp_transport_controller_send_interface.h"
-#include "call/simulated_network.h"
 #include "call/video_receive_stream.h"
 #include "call/video_send_stream.h"
 #include "rtc_base/copy_on_write_buffer.h"
 #include "rtc_base/network/sent_packet.h"
 #include "system_wrappers/include/clock.h"
+#include "test/network/simulated_network.h"
 #include "video/config/video_encoder_config.h"
 
 namespace webrtc {

call/fake_network_pipe_unittest.cc

@@ -15,7 +15,6 @@
 
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
-#include "call/simulated_network.h"
 #include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
 #include "modules/rtp_rtcp/source/rtp_header_extensions.h"
 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
@@ -23,6 +22,7 @@
 #include "system_wrappers/include/clock.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/network/simulated_network.h"
 
 using ::testing::_;
 using ::testing::Property;

call/rampup_tests.h

@@ -23,7 +23,6 @@
 #include "api/test/metrics/metric.h"
 #include "api/test/simulated_network.h"
 #include "call/call.h"
-#include "call/simulated_network.h"
 #include "rtc_base/event.h"
 #include "rtc_base/task_utils/repeating_task.h"
 #include "test/call_test.h"

call/simulated_network.h

@@ -10,125 +10,8 @@
 #ifndef CALL_SIMULATED_NETWORK_H_
 #define CALL_SIMULATED_NETWORK_H_
 
-#include <stdint.h>
+// TODO(bugs.webrtc.org/14525): Remove once downstream projects does not use
-
+// it.
-#include <deque>
+#include "test/network/simulated_network.h"
-#include <queue>
-#include <vector>
-
-#include "absl/types/optional.h"
-#include "api/sequence_checker.h"
-#include "api/test/simulated_network.h"
-#include "api/units/data_size.h"
-#include "api/units/timestamp.h"
-#include "rtc_base/race_checker.h"
-#include "rtc_base/random.h"
-#include "rtc_base/synchronization/mutex.h"
-#include "rtc_base/thread_annotations.h"
-
-namespace webrtc {
-
-// Class simulating a network link.
-//
-// This is a basic implementation of NetworkBehaviorInterface that supports:
-// - Packet loss
-// - Capacity delay
-// - Extra delay with or without packets reorder
-// - Packet overhead
-// - Queue max capacity
-class RTC_EXPORT SimulatedNetwork : public SimulatedNetworkInterface {
- public:
-  using Config = BuiltInNetworkBehaviorConfig;
-  explicit SimulatedNetwork(Config config, uint64_t random_seed = 1);
-  ~SimulatedNetwork() override;
-
-  // Sets a new configuration. This will affect packets that will be sent with
-  // EnqueuePacket but also packets in the network that have not left the
-  // network emulation. Packets that are ready to be retrieved by
-  // DequeueDeliverablePackets are not affected by the new configuration.
-  // TODO(bugs.webrtc.org/14525): Fix SetConfig and make it apply only to the
-  // part of the packet that is currently being sent (instead of applying to
-  // all of it).
-  void SetConfig(const Config& config) override;
-  void UpdateConfig(std::function<void(BuiltInNetworkBehaviorConfig*)>
-                        config_modifier) override;
-  void PauseTransmissionUntil(int64_t until_us) override;
-
-  // NetworkBehaviorInterface
-  bool EnqueuePacket(PacketInFlightInfo packet) override;
-  std::vector<PacketDeliveryInfo> DequeueDeliverablePackets(
-      int64_t receive_time_us) override;
-
-  absl::optional<int64_t> NextDeliveryTimeUs() const override;
-
- private:
-  struct PacketInfo {
-    PacketInFlightInfo packet;
-    // Time when the packet has left (or will leave) the network.
-    int64_t arrival_time_us;
-  };
-  // Contains current configuration state.
-  struct ConfigState {
-    // Static link configuration.
-    Config config;
-    // The probability to drop the packet if we are currently dropping a
-    // burst of packet
-    double prob_loss_bursting;
-    // The probability to drop a burst of packets.
-    double prob_start_bursting;
-    // Used for temporary delay spikes.
-    int64_t pause_transmission_until_us = 0;
-  };
-
-  // Moves packets from capacity- to delay link.
-  void UpdateCapacityQueue(ConfigState state, int64_t time_now_us)
-      RTC_RUN_ON(&process_checker_);
-  ConfigState GetConfigState() const;
-
-  mutable Mutex config_lock_;
-
-  // Guards the data structures involved in delay and loss processing, such as
-  // the packet queues.
-  rtc::RaceChecker process_checker_;
-  // Models the capacity of the network by rejecting packets if the queue is
-  // full and keeping them in the queue until they are ready to exit (according
-  // to the link capacity, which cannot be violated, e.g. a 1 kbps link will
-  // only be able to deliver 1000 bits per second).
-  //
-  // Invariant:
-  // The head of the `capacity_link_` has arrival_time_us correctly set to the
-  // time when the packet is supposed to be delivered (without accounting
-  // potential packet loss or potential extra delay and without accounting for a
-  // new configuration of the network, which requires a re-computation of the
-  // arrival_time_us).
-  std::queue<PacketInfo> capacity_link_ RTC_GUARDED_BY(process_checker_);
-  // Models the extra delay of the network (see `queue_delay_ms`
-  // and `delay_standard_deviation_ms` in BuiltInNetworkBehaviorConfig), packets
-  // in the `delay_link_` have technically already left the network and don't
-  // use its capacity but they are not delivered yet.
-  std::deque<PacketInfo> delay_link_ RTC_GUARDED_BY(process_checker_);
-  // Represents the next moment in time when the network is supposed to deliver
-  // packets to the client (either by pulling them from `delay_link_` or
-  // `capacity_link_` or both).
-  absl::optional<int64_t> next_process_time_us_
-      RTC_GUARDED_BY(process_checker_);
-
-  ConfigState config_state_ RTC_GUARDED_BY(config_lock_);
-
-  Random random_ RTC_GUARDED_BY(process_checker_);
-  // Are we currently dropping a burst of packets?
-  bool bursting_;
-
-  // The send time of the last enqueued packet, this is only used to check that
-  // the send time of enqueued packets is monotonically increasing.
-  int64_t last_enqueue_time_us_;
-
-  // The last time a packet left the capacity_link_ (used to enforce
-  // the capacity of the link and avoid packets starts to get sent before
-  // the link it free).
-  int64_t last_capacity_link_exit_time_;
-};
-
-}  // namespace webrtc
 
 #endif  // CALL_SIMULATED_NETWORK_H_

net/dcsctp/socket/BUILD.gn

@@ -236,7 +236,6 @@ if (rtc_include_tests) {
       "../../../api/task_queue",
       "../../../api/task_queue:pending_task_safety_flag",
       "../../../api/units:time_delta",
-      "../../../call:simulated_network",
       "../../../rtc_base:checks",
       "../../../rtc_base:copy_on_write_buffer",
       "../../../rtc_base:gunit_helpers",

net/dcsctp/socket/dcsctp_socket_network_test.cc

@@ -23,7 +23,6 @@
 #include "api/test/create_network_emulation_manager.h"
 #include "api/test/network_emulation_manager.h"
 #include "api/units/time_delta.h"
-#include "call/simulated_network.h"
 #include "net/dcsctp/public/dcsctp_options.h"
 #include "net/dcsctp/public/dcsctp_socket.h"
 #include "net/dcsctp/public/types.h"

pc/BUILD.gn

@@ -2833,6 +2833,7 @@ if (rtc_include_tests && !build_with_chromium) {
     sources = [ "test/svc_e2e_tests.cc" ]
     data = svc_tests_resources
     deps = [
+      "..//test/network:simulated_network",
       "../api:create_network_emulation_manager",
       "../api:create_peer_connection_quality_test_frame_generator",
       "../api:create_peerconnection_quality_test_fixture",
@@ -2848,7 +2849,6 @@ if (rtc_include_tests && !build_with_chromium) {
       "../api/test/pclf:media_quality_test_params",
       "../api/test/pclf:peer_configurer",
       "../api/video_codecs:video_codecs_api",
-      "../call:simulated_network",
       "../modules/video_coding:webrtc_vp9",
       "../modules/video_coding/svc:scalability_mode_util",
       "../rtc_base/containers:flat_map",

pc/test/svc_e2e_tests.cc

@@ -27,7 +27,6 @@
 #include "api/test/simulated_network.h"
 #include "api/test/time_controller.h"
 #include "api/video_codecs/vp9_profile.h"
-#include "call/simulated_network.h"
 #include "modules/video_coding/codecs/vp9/include/vp9.h"
 #include "modules/video_coding/svc/scalability_mode_util.h"
 #include "rtc_base/containers/flat_map.h"
@@ -35,6 +34,7 @@
 #include "test/field_trial.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/network/simulated_network.h"
 #include "test/pc/e2e/analyzer/video/default_video_quality_analyzer.h"
 #include "test/pc/e2e/network_quality_metrics_reporter.h"
 #include "test/testsupport/file_utils.h"

rtc_tools/BUILD.gn

@@ -201,6 +201,7 @@ if (!is_component_build) {
       "..//api/video_codecs:video_encoder_factory_template_libvpx_vp8_adapter",
       "..//api/video_codecs:video_encoder_factory_template_libvpx_vp9_adapter",
       "..//api/video_codecs:video_encoder_factory_template_open_h264_adapter",
+      "..//test/network:simulated_network",
       "../api:create_frame_generator",
       "../api:rtp_parameters",
       "../api:transport_api",
@@ -213,7 +214,6 @@ if (!is_component_build) {
       "../call:fake_network",
       "../call:rtp_interfaces",
       "../call:rtp_sender",
-      "../call:simulated_network",
       "../call:simulated_packet_receiver",
       "../call:video_stream_api",
       "../media:media_constants",

test/BUILD.gn

@@ -1266,6 +1266,7 @@ if (!build_with_chromium) {
       ":test_video_capturer",
       ":video_test_common",
       ":video_test_constants",
+      "..//test/network:simulated_network",
       "../api:array_view",
       "../api:create_frame_generator",
       "../api:frame_generator_api",
@@ -1290,7 +1291,6 @@ if (!build_with_chromium) {
       "../call",
       "../call:call_interfaces",
       "../call:fake_network",
-      "../call:simulated_network",
       "../call:simulated_packet_receiver",
       "../call:video_stream_api",
       "../modules/audio_device:test_audio_device_module",

test/call_test.cc

@@ -23,13 +23,13 @@
 #include "api/video/builtin_video_bitrate_allocator_factory.h"
 #include "call/fake_network_pipe.h"
 #include "call/packet_receiver.h"
-#include "call/simulated_network.h"
 #include "modules/audio_device/include/test_audio_device.h"
 #include "modules/audio_mixer/audio_mixer_impl.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/event.h"
 #include "rtc_base/task_queue_for_test.h"
 #include "test/fake_encoder.h"
+#include "test/network/simulated_network.h"
 #include "test/rtp_rtcp_observer.h"
 #include "test/testsupport/file_utils.h"
 #include "test/video_test_constants.h"

test/network/BUILD.gn

@@ -38,6 +38,7 @@ rtc_library("emulated_network") {
     "traffic_route.h",
   ]
   deps = [
+    "../..//test/network:simulated_network",
     "../../api:array_view",
     "../../api:field_trials_view",
     "../../api:network_emulation_manager_api",
@@ -55,7 +56,6 @@ rtc_library("emulated_network") {
     "../../api/units:data_size",
     "../../api/units:time_delta",
     "../../api/units:timestamp",
-    "../../call:simulated_network",
     "../../p2p:basic_packet_socket_factory",
     "../../p2p:p2p_server_utils",
     "../../p2p:rtc_p2p",
@@ -101,9 +101,9 @@ rtc_library("network_emulation_unittest") {
   deps = [
     ":emulated_network",
     "../:test_support",
+    "../..//test/network:simulated_network",
     "../../api:simulated_network_api",
     "../../api/units:time_delta",
-    "../../call:simulated_network",
     "../../rtc_base:gunit_helpers",
     "../../rtc_base:logging",
     "../../rtc_base:rtc_event",
@@ -119,6 +119,7 @@ if (rtc_include_tests && !build_with_chromium) {
     deps = [
       ":emulated_network",
       "../:test_support",
+      "../..//test/network:simulated_network",
       "../../api:enable_media_with_defaults",
       "../../api:libjingle_peerconnection_api",
       "../../api:scoped_refptr",
@@ -126,7 +127,6 @@ if (rtc_include_tests && !build_with_chromium) {
       "../../api/rtc_event_log:rtc_event_log_factory",
       "../../api/task_queue:default_task_queue_factory",
       "../../api/transport:field_trial_based_config",
-      "../../call:simulated_network",
       "../../media:rtc_audio_video",
       "../../modules/audio_device:test_audio_device_module",
       "../../p2p:basic_packet_socket_factory",
@@ -148,9 +148,9 @@ rtc_library("cross_traffic_unittest") {
   deps = [
     ":emulated_network",
     "../:test_support",
+    "../..//test/network:simulated_network",
     "../../api:network_emulation_manager_api",
     "../../api:simulated_network_api",
-    "../../call:simulated_network",
     "../../rtc_base:logging",
     "../../rtc_base:network_constants",
     "../../rtc_base:rtc_event",
@@ -171,9 +171,9 @@ if (rtc_include_tests) {
     ]
     deps = [
       ":emulated_network",
+      "../..//test/network:simulated_network",
       "../../api/transport:network_control",
       "../../api/transport:test_feedback_generator_interface",
-      "../../call:simulated_network",
       "../../rtc_base:checks",
       "../time_controller",
     ]
@@ -197,7 +197,46 @@ if (rtc_include_tests) {
         ":feedback_generator_unittest",
         ":network_emulation_pc_unittest",
         ":network_emulation_unittest",
+        ":simulated_network_unittest",
       ]
     }
   }
 }
+
+rtc_library("simulated_network") {
+  sources = [
+    "simulated_network.cc",
+    "simulated_network.h",
+  ]
+  deps = [
+    "../../api:sequence_checker",
+    "../../api:simulated_network_api",
+    "../../api/units:data_rate",
+    "../../api/units:data_size",
+    "../../api/units:time_delta",
+    "../../api/units:timestamp",
+    "../../rtc_base:checks",
+    "../../rtc_base:macromagic",
+    "../../rtc_base:race_checker",
+    "../../rtc_base:random",
+    "../../rtc_base/synchronization:mutex",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+if (rtc_include_tests) {
+  rtc_library("simulated_network_unittest") {
+    testonly = true
+    sources = [ "simulated_network_unittest.cc" ]
+    deps = [
+      ":simulated_network",
+      "../:test_support",
+      "../../api:simulated_network_api",
+      "../../api/units:data_rate",
+      "../../api/units:data_size",
+      "../../api/units:time_delta",
+      "//testing/gtest",
+    ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/algorithm:container" ]
+  }
+}

test/network/cross_traffic_unittest.cc

@@ -19,13 +19,13 @@
 #include "absl/types/optional.h"
 #include "api/test/network_emulation_manager.h"
 #include "api/test/simulated_network.h"
-#include "call/simulated_network.h"
 #include "rtc_base/event.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/network_constants.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
 #include "test/network/network_emulation_manager.h"
+#include "test/network/simulated_network.h"
 #include "test/network/traffic_route.h"
 #include "test/time_controller/simulated_time_controller.h"
 

test/network/feedback_generator.h

@@ -17,9 +17,9 @@
 
 #include "api/transport/network_types.h"
 #include "api/transport/test/feedback_generator_interface.h"
-#include "call/simulated_network.h"
 #include "test/network/network_emulation.h"
 #include "test/network/network_emulation_manager.h"
+#include "test/network/simulated_network.h"
 #include "test/time_controller/simulated_time_controller.h"
 
 namespace webrtc {

test/network/network_emulation_manager.cc

@@ -16,8 +16,8 @@
 #include "api/field_trials_view.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
-#include "call/simulated_network.h"
 #include "test/network/emulated_turn_server.h"
+#include "test/network/simulated_network.h"
 #include "test/network/traffic_route.h"
 #include "test/time_controller/real_time_controller.h"
 #include "test/time_controller/simulated_time_controller.h"

test/network/network_emulation_pc_unittest.cc

@@ -17,7 +17,6 @@
 #include "api/scoped_refptr.h"
 #include "api/task_queue/default_task_queue_factory.h"
 #include "api/transport/field_trial_based_config.h"
-#include "call/simulated_network.h"
 #include "media/engine/webrtc_media_engine.h"
 #include "modules/audio_device/include/test_audio_device.h"
 #include "p2p/base/basic_packet_socket_factory.h"
@@ -30,6 +29,7 @@
 #include "test/gtest.h"
 #include "test/network/network_emulation.h"
 #include "test/network/network_emulation_manager.h"
+#include "test/network/simulated_network.h"
 
 namespace webrtc {
 namespace test {

test/network/network_emulation_unittest.cc

@@ -16,7 +16,6 @@
 
 #include "api/test/simulated_network.h"
 #include "api/units/time_delta.h"
-#include "call/simulated_network.h"
 #include "rtc_base/event.h"
 #include "rtc_base/gunit.h"
 #include "rtc_base/synchronization/mutex.h"
@@ -24,6 +23,7 @@
 #include "test/gmock.h"
 #include "test/gtest.h"
 #include "test/network/network_emulation_manager.h"
+#include "test/network/simulated_network.h"
 
 namespace webrtc {
 namespace test {

test/network/simulated_network.cc

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "call/simulated_network.h"
+#include "test/network/simulated_network.h"
 
 #include <algorithm>
 #include <cmath>

test/network/simulated_network.h

@@ -0,0 +1,134 @@
+/*
+ *  Copyright 2018 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.
+ */
+#ifndef TEST_NETWORK_SIMULATED_NETWORK_H_
+#define TEST_NETWORK_SIMULATED_NETWORK_H_
+
+#include <stdint.h>
+
+#include <deque>
+#include <queue>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/sequence_checker.h"
+#include "api/test/simulated_network.h"
+#include "api/units/data_size.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/race_checker.h"
+#include "rtc_base/random.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+
+// Class simulating a network link.
+//
+// This is a basic implementation of NetworkBehaviorInterface that supports:
+// - Packet loss
+// - Capacity delay
+// - Extra delay with or without packets reorder
+// - Packet overhead
+// - Queue max capacity
+class RTC_EXPORT SimulatedNetwork : public SimulatedNetworkInterface {
+ public:
+  using Config = BuiltInNetworkBehaviorConfig;
+  explicit SimulatedNetwork(Config config, uint64_t random_seed = 1);
+  ~SimulatedNetwork() override;
+
+  // Sets a new configuration. This will affect packets that will be sent with
+  // EnqueuePacket but also packets in the network that have not left the
+  // network emulation. Packets that are ready to be retrieved by
+  // DequeueDeliverablePackets are not affected by the new configuration.
+  // TODO(bugs.webrtc.org/14525): Fix SetConfig and make it apply only to the
+  // part of the packet that is currently being sent (instead of applying to
+  // all of it).
+  void SetConfig(const Config& config) override;
+  void UpdateConfig(std::function<void(BuiltInNetworkBehaviorConfig*)>
+                        config_modifier) override;
+  void PauseTransmissionUntil(int64_t until_us) override;
+
+  // NetworkBehaviorInterface
+  bool EnqueuePacket(PacketInFlightInfo packet) override;
+  std::vector<PacketDeliveryInfo> DequeueDeliverablePackets(
+      int64_t receive_time_us) override;
+
+  absl::optional<int64_t> NextDeliveryTimeUs() const override;
+
+ private:
+  struct PacketInfo {
+    PacketInFlightInfo packet;
+    // Time when the packet has left (or will leave) the network.
+    int64_t arrival_time_us;
+  };
+  // Contains current configuration state.
+  struct ConfigState {
+    // Static link configuration.
+    Config config;
+    // The probability to drop the packet if we are currently dropping a
+    // burst of packet
+    double prob_loss_bursting;
+    // The probability to drop a burst of packets.
+    double prob_start_bursting;
+    // Used for temporary delay spikes.
+    int64_t pause_transmission_until_us = 0;
+  };
+
+  // Moves packets from capacity- to delay link.
+  void UpdateCapacityQueue(ConfigState state, int64_t time_now_us)
+      RTC_RUN_ON(&process_checker_);
+  ConfigState GetConfigState() const;
+
+  mutable Mutex config_lock_;
+
+  // Guards the data structures involved in delay and loss processing, such as
+  // the packet queues.
+  rtc::RaceChecker process_checker_;
+  // Models the capacity of the network by rejecting packets if the queue is
+  // full and keeping them in the queue until they are ready to exit (according
+  // to the link capacity, which cannot be violated, e.g. a 1 kbps link will
+  // only be able to deliver 1000 bits per second).
+  //
+  // Invariant:
+  // The head of the `capacity_link_` has arrival_time_us correctly set to the
+  // time when the packet is supposed to be delivered (without accounting
+  // potential packet loss or potential extra delay and without accounting for a
+  // new configuration of the network, which requires a re-computation of the
+  // arrival_time_us).
+  std::queue<PacketInfo> capacity_link_ RTC_GUARDED_BY(process_checker_);
+  // Models the extra delay of the network (see `queue_delay_ms`
+  // and `delay_standard_deviation_ms` in BuiltInNetworkBehaviorConfig), packets
+  // in the `delay_link_` have technically already left the network and don't
+  // use its capacity but they are not delivered yet.
+  std::deque<PacketInfo> delay_link_ RTC_GUARDED_BY(process_checker_);
+  // Represents the next moment in time when the network is supposed to deliver
+  // packets to the client (either by pulling them from `delay_link_` or
+  // `capacity_link_` or both).
+  absl::optional<int64_t> next_process_time_us_
+      RTC_GUARDED_BY(process_checker_);
+
+  ConfigState config_state_ RTC_GUARDED_BY(config_lock_);
+
+  Random random_ RTC_GUARDED_BY(process_checker_);
+  // Are we currently dropping a burst of packets?
+  bool bursting_;
+
+  // The send time of the last enqueued packet, this is only used to check that
+  // the send time of enqueued packets is monotonically increasing.
+  int64_t last_enqueue_time_us_;
+
+  // The last time a packet left the capacity_link_ (used to enforce
+  // the capacity of the link and avoid packets starts to get sent before
+  // the link it free).
+  int64_t last_capacity_link_exit_time_;
+};
+
+}  // namespace webrtc
+
+#endif  // TEST_NETWORK_SIMULATED_NETWORK_H_

test/network/simulated_network_unittest.cc

@@ -7,7 +7,7 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#include "call/simulated_network.h"
+#include "test/network/simulated_network.h"
 
 #include <algorithm>
 #include <map>

test/pc/e2e/BUILD.gn

@@ -287,6 +287,7 @@ if (!build_with_chromium) {
         ":default_audio_quality_analyzer",
         ":network_quality_metrics_reporter",
         ":stats_based_network_quality_metrics_reporter",
+        "../../..//test/network:simulated_network",
         "../../../api:create_network_emulation_manager",
         "../../../api:create_peer_connection_quality_test_frame_generator",
         "../../../api:create_peerconnection_quality_test_fixture",
@@ -304,7 +305,6 @@ if (!build_with_chromium) {
         "../../../api/test/pclf:peer_configurer",
         "../../../api/video_codecs:builtin_video_decoder_factory",
         "../../../api/video_codecs:builtin_video_encoder_factory",
-        "../../../call:simulated_network",
         "../../../media:rtc_audio_video",
         "../../../modules/audio_device:audio_device_impl",
         "../../../p2p:rtc_p2p",

test/pc/e2e/peer_connection_e2e_smoke_test.cc

@@ -22,10 +22,10 @@
 #include "api/test/pclf/media_quality_test_params.h"
 #include "api/test/pclf/peer_configurer.h"
 #include "api/test/peerconnection_quality_test_fixture.h"
-#include "call/simulated_network.h"
 #include "system_wrappers/include/field_trial.h"
 #include "test/field_trial.h"
 #include "test/gtest.h"
+#include "test/network/simulated_network.h"
 #include "test/pc/e2e/analyzer/audio/default_audio_quality_analyzer.h"
 #include "test/pc/e2e/analyzer/video/default_video_quality_analyzer.h"
 #include "test/pc/e2e/analyzer/video/default_video_quality_analyzer_shared_objects.h"

test/scenario/BUILD.gn

@@ -77,6 +77,7 @@ if (rtc_include_tests && !build_with_chromium) {
       "../:test_common",
       "../:test_support",
       "../:video_test_common",
+      "../..//test/network:simulated_network",
       "../../api:array_view",
       "../../api:create_frame_generator",
       "../../api:fec_controller_api",
@@ -110,7 +111,6 @@ if (rtc_include_tests && !build_with_chromium) {
       "../../call",
       "../../call:call_interfaces",
       "../../call:rtp_sender",
-      "../../call:simulated_network",
       "../../call:video_stream_api",
       "../../common_video",
       "../../media:media_constants",

test/scenario/network_node.h

@@ -19,10 +19,10 @@
 #include "api/call/transport.h"
 #include "api/units/timestamp.h"
 #include "call/call.h"
-#include "call/simulated_network.h"
 #include "rtc_base/copy_on_write_buffer.h"
 #include "rtc_base/synchronization/mutex.h"
 #include "test/network/network_emulation.h"
+#include "test/network/simulated_network.h"
 #include "test/scenario/column_printer.h"
 #include "test/scenario/scenario_config.h"
 

video/BUILD.gn

@@ -513,6 +513,7 @@ if (rtc_include_tests) {
       ]
       deps = [
         ":frame_dumping_decoder",
+        "..//test/network:simulated_network",
         "../api:create_frame_generator",
         "../api:fec_controller_api",
         "../api:frame_generator_api",
@@ -534,7 +535,6 @@ if (rtc_include_tests) {
         "../api/video:video_rtp_headers",
         "../api/video_codecs:video_codecs_api",
         "../call:fake_network",
-        "../call:simulated_network",
         "../common_video",
         "../media:media_constants",
         "../media:rtc_audio_video",
@@ -623,6 +623,7 @@ if (rtc_include_tests) {
 
       sources = [ "pc_full_stack_tests.cc" ]
       deps = [
+        "..//test/network:simulated_network",
         "../api:create_network_emulation_manager",
         "../api:create_peer_connection_quality_test_frame_generator",
         "../api:create_peerconnection_quality_test_fixture",
@@ -637,7 +638,6 @@ if (rtc_include_tests) {
         "../api/test/pclf:media_quality_test_params",
         "../api/test/pclf:peer_configurer",
         "../api/video_codecs:video_codecs_api",
-        "../call:simulated_network",
         "../modules/video_coding:webrtc_vp9",
         "../system_wrappers:field_trial",
         "../test:field_trial",
@@ -820,6 +820,7 @@ if (rtc_include_tests) {
       ":video_stream_buffer_controller",
       ":video_stream_encoder_impl",
       ":video_stream_encoder_interface",
+      "..//test/network:simulated_network",
       "../api:array_view",
       "../api:bitrate_allocation",
       "../api:create_frame_generator",
@@ -879,7 +880,6 @@ if (rtc_include_tests) {
       "../call:rtp_interfaces",
       "../call:rtp_receiver",
       "../call:rtp_sender",
-      "../call:simulated_network",
       "../call:simulated_packet_receiver",
       "../call:video_stream_api",
       "../call/adaptation:resource_adaptation",

video/end_to_end_tests/bandwidth_tests.cc

@@ -16,7 +16,6 @@
 #include "api/video/builtin_video_bitrate_allocator_factory.h"
 #include "api/video/video_bitrate_allocation.h"
 #include "call/fake_network_pipe.h"
-#include "call/simulated_network.h"
 #include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
 #include "rtc_base/rate_limiter.h"
 #include "rtc_base/synchronization/mutex.h"
@@ -26,6 +25,7 @@
 #include "test/fake_encoder.h"
 #include "test/field_trial.h"
 #include "test/gtest.h"
+#include "test/network/simulated_network.h"
 #include "test/rtcp_packet_parser.h"
 #include "test/rtp_rtcp_observer.h"
 #include "test/video_encoder_proxy_factory.h"

video/end_to_end_tests/call_operation_tests.cc

@@ -14,13 +14,13 @@
 #include "api/test/frame_generator_interface.h"
 #include "api/test/simulated_network.h"
 #include "call/fake_network_pipe.h"
-#include "call/simulated_network.h"
 #include "rtc_base/task_queue_for_test.h"
 #include "system_wrappers/include/sleep.h"
 #include "test/call_test.h"
 #include "test/field_trial.h"
 #include "test/frame_forwarder.h"
 #include "test/gtest.h"
+#include "test/network/simulated_network.h"
 #include "test/null_transport.h"
 #include "test/video_test_constants.h"
 

video/end_to_end_tests/extended_reports_tests.cc

@@ -24,7 +24,6 @@
 #include "call/call.h"
 #include "call/fake_network_pipe.h"
 #include "call/rtp_config.h"
-#include "call/simulated_network.h"
 #include "call/simulated_packet_receiver.h"
 #include "call/video_receive_stream.h"
 #include "call/video_send_stream.h"
@@ -37,6 +36,7 @@
 #include "test/call_test.h"
 #include "test/field_trial.h"
 #include "test/gtest.h"
+#include "test/network/simulated_network.h"
 #include "test/rtcp_packet_parser.h"
 #include "test/rtp_rtcp_observer.h"
 #include "test/video_test_constants.h"

video/end_to_end_tests/fec_tests.cc

@@ -14,7 +14,6 @@
 #include "api/test/simulated_network.h"
 #include "api/test/video/function_video_encoder_factory.h"
 #include "call/fake_network_pipe.h"
-#include "call/simulated_network.h"
 #include "media/engine/internal_decoder_factory.h"
 #include "modules/include/module_common_types_public.h"
 #include "modules/rtp_rtcp/source/byte_io.h"
@@ -25,6 +24,7 @@
 #include "test/field_trial.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/network/simulated_network.h"
 #include "test/rtcp_packet_parser.h"
 #include "test/video_test_constants.h"
 

video/end_to_end_tests/multi_codec_receive_tests.cc

@@ -13,7 +13,6 @@
 #include "api/test/simulated_network.h"
 #include "api/test/video/function_video_encoder_factory.h"
 #include "call/fake_network_pipe.h"
-#include "call/simulated_network.h"
 #include "modules/include/module_common_types_public.h"
 #include "modules/rtp_rtcp/source/rtp_packet.h"
 #include "modules/video_coding/codecs/h264/include/h264.h"
@@ -24,6 +23,7 @@
 #include "test/call_test.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/network/simulated_network.h"
 #include "test/video_test_constants.h"
 
 using ::testing::Contains;

video/end_to_end_tests/multi_stream_tester.cc

@@ -22,12 +22,12 @@
 #include "api/test/video/function_video_encoder_factory.h"
 #include "api/video/builtin_video_bitrate_allocator_factory.h"
 #include "call/fake_network_pipe.h"
-#include "call/simulated_network.h"
 #include "media/engine/internal_decoder_factory.h"
 #include "modules/video_coding/codecs/vp8/include/vp8.h"
 #include "rtc_base/task_queue_for_test.h"
 #include "test/call_test.h"
 #include "test/encoder_settings.h"
+#include "test/network/simulated_network.h"
 #include "test/video_test_constants.h"
 
 namespace webrtc {

video/end_to_end_tests/network_state_tests.cc

@@ -17,7 +17,6 @@
 #include "api/test/simulated_network.h"
 #include "api/video_codecs/video_encoder.h"
 #include "call/fake_network_pipe.h"
-#include "call/simulated_network.h"
 #include "modules/rtp_rtcp/source/rtp_packet.h"
 #include "rtc_base/synchronization/mutex.h"
 #include "rtc_base/task_queue_for_test.h"
@@ -25,6 +24,7 @@
 #include "test/call_test.h"
 #include "test/fake_encoder.h"
 #include "test/gtest.h"
+#include "test/network/simulated_network.h"
 #include "test/video_encoder_proxy_factory.h"
 
 namespace webrtc {

video/end_to_end_tests/retransmission_tests.cc

@@ -16,7 +16,6 @@
 #include "api/test/video/function_video_encoder_factory.h"
 #include "api/units/time_delta.h"
 #include "call/fake_network_pipe.h"
-#include "call/simulated_network.h"
 #include "modules/rtp_rtcp/source/rtp_packet.h"
 #include "modules/video_coding/codecs/vp8/include/vp8.h"
 #include "rtc_base/event.h"
@@ -25,6 +24,7 @@
 #include "test/call_test.h"
 #include "test/field_trial.h"
 #include "test/gtest.h"
+#include "test/network/simulated_network.h"
 #include "test/rtcp_packet_parser.h"
 #include "test/video_test_constants.h"
 

video/end_to_end_tests/rtp_rtcp_tests.cc

@@ -12,7 +12,6 @@
 
 #include "api/test/simulated_network.h"
 #include "call/fake_network_pipe.h"
-#include "call/simulated_network.h"
 #include "modules/include/module_common_types_public.h"
 #include "modules/rtp_rtcp/source/rtp_packet.h"
 #include "modules/video_coding/codecs/vp8/include/vp8.h"
@@ -21,6 +20,7 @@
 #include "rtc_base/task_queue_for_test.h"
 #include "test/call_test.h"
 #include "test/gtest.h"
+#include "test/network/simulated_network.h"
 #include "test/rtcp_packet_parser.h"
 #include "test/video_test_constants.h"
 

video/end_to_end_tests/ssrc_tests.cc

@@ -13,13 +13,13 @@
 #include "api/test/simulated_network.h"
 #include "call/fake_network_pipe.h"
 #include "call/packet_receiver.h"
-#include "call/simulated_network.h"
 #include "modules/rtp_rtcp/source/rtp_packet.h"
 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
 #include "modules/rtp_rtcp/source/rtp_util.h"
 #include "rtc_base/task_queue_for_test.h"
 #include "test/call_test.h"
 #include "test/gtest.h"
+#include "test/network/simulated_network.h"
 #include "test/rtcp_packet_parser.h"
 #include "test/video_test_constants.h"
 

video/end_to_end_tests/stats_tests.cc

@@ -16,7 +16,6 @@
 #include "api/test/simulated_network.h"
 #include "api/test/video/function_video_encoder_factory.h"
 #include "call/fake_network_pipe.h"
-#include "call/simulated_network.h"
 #include "modules/rtp_rtcp/source/rtp_packet.h"
 #include "modules/video_coding/include/video_coding_defines.h"
 #include "rtc_base/strings/string_builder.h"
@@ -27,6 +26,7 @@
 #include "test/call_test.h"
 #include "test/fake_encoder.h"
 #include "test/gtest.h"
+#include "test/network/simulated_network.h"
 #include "test/rtcp_packet_parser.h"
 #include "test/video_test_constants.h"
 

video/end_to_end_tests/transport_feedback_tests.cc

@@ -16,7 +16,6 @@
 #include "api/units/time_delta.h"
 #include "call/call.h"
 #include "call/fake_network_pipe.h"
-#include "call/simulated_network.h"
 #include "modules/rtp_rtcp/source/byte_io.h"
 #include "modules/rtp_rtcp/source/rtp_header_extensions.h"
 #include "modules/rtp_rtcp/source/rtp_packet.h"
@@ -25,6 +24,7 @@
 #include "test/call_test.h"
 #include "test/field_trial.h"
 #include "test/gtest.h"
+#include "test/network/simulated_network.h"
 #include "test/rtcp_packet_parser.h"
 #include "test/video_test_constants.h"
 #include "video/end_to_end_tests/multi_stream_tester.h"

video/pc_full_stack_tests.cc

@@ -26,11 +26,11 @@
 #include "api/test/simulated_network.h"
 #include "api/test/time_controller.h"
 #include "api/video_codecs/vp9_profile.h"
-#include "call/simulated_network.h"
 #include "modules/video_coding/codecs/vp9/include/vp9.h"
 #include "system_wrappers/include/field_trial.h"
 #include "test/field_trial.h"
 #include "test/gtest.h"
+#include "test/network/simulated_network.h"
 #include "test/pc/e2e/network_quality_metrics_reporter.h"
 #include "test/testsupport/file_utils.h"
 

video/picture_id_tests.cc

@@ -13,7 +13,6 @@
 #include "api/test/simulated_network.h"
 #include "api/test/video/function_video_encoder_factory.h"
 #include "call/fake_network_pipe.h"
-#include "call/simulated_network.h"
 #include "media/engine/internal_encoder_factory.h"
 #include "media/engine/simulcast_encoder_adapter.h"
 #include "modules/rtp_rtcp/source/create_video_rtp_depacketizer.h"
@@ -25,6 +24,7 @@
 #include "rtc_base/synchronization/mutex.h"
 #include "rtc_base/task_queue_for_test.h"
 #include "test/call_test.h"
+#include "test/network/simulated_network.h"
 #include "test/video_test_constants.h"
 
 namespace webrtc {

video/video_quality_test.cc

@@ -31,7 +31,6 @@
 #include "api/video/builtin_video_bitrate_allocator_factory.h"
 #include "api/video_codecs/video_encoder.h"
 #include "call/fake_network_pipe.h"
-#include "call/simulated_network.h"
 #include "media/base/media_constants.h"
 #include "media/engine/adm_helpers.h"
 #include "media/engine/fake_video_codec_factory.h"
@@ -45,6 +44,7 @@
 #include "modules/video_coding/utility/ivf_file_writer.h"
 #include "rtc_base/strings/string_builder.h"
 #include "rtc_base/task_queue_for_test.h"
+#include "test/network/simulated_network.h"
 #include "test/platform_video_capturer.h"
 #include "test/test_flags.h"
 #include "test/testsupport/file_utils.h"

video/video_send_stream_tests.cc

@@ -26,7 +26,6 @@
 #include "call/call.h"
 #include "call/fake_network_pipe.h"
 #include "call/rtp_transport_controller_send.h"
-#include "call/simulated_network.h"
 #include "call/video_send_stream.h"
 #include "media/engine/internal_encoder_factory.h"
 #include "media/engine/simulcast_encoder_adapter.h"
@@ -66,6 +65,7 @@
 #include "test/frame_utils.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
+#include "test/network/simulated_network.h"
 #include "test/null_transport.h"
 #include "test/rtcp_packet_parser.h"
 #include "test/rtp_rtcp_observer.h"