Add support for time-varying constraints in DegradedCall.

The fake network configs are now specified using just two field trials:
WebRTC-FakeNetworkSendConfig and WebRTC-FakeNetworkReceiveConfig.
Both of them have the following parameters from
BuiltInNetworkBehaviorConfig:

* queue_length_packets  // Queue length in number of packets.
* queue_delay_ms  // Delay in addition to capacity induced delay.
* delay_standard_deviation_ms   // Standard deviation of the extra delay.
* link_capacity_kbps  // Link capacity in kbps.
* loss_percent  // Random packet loss.
* allow_reordering  // If packets are allowed to be reordered.
* avg_burst_loss_length  // The average length of a burst of lost packets.
* packet_overhead  // Additional bytes to add to packet size.
* codel_active_queue_management  // Enable CoDel active queue management.

Plus:
* duration  // For how long to use this config before progressing.

Example:
WebRTC-FakeNetworkSendConfig/queue_delay_ms:66|1,loss_percent:1|0,link_capacity_kbps:200|10000,queue_length_packets:10|100,duration:15s|45s/
This creates two configs:
1. For 15s, apply 66ms delay, 1% loss, 200kbps bandwidth, 10 packet queue size
2. For 45s, apply 1ms delay, 0% loss, 10Mbps bandwidth, 100 packets queue size
(then repeat)

Bug: webrtc:13655
Change-Id: I0524f572de480731df4d414724203772182c628b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251043
Reviewed-by: Stefan Holmer <holmer@google.com>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35952}

call/call_factory.cc

@@ -15,26 +15,34 @@
 #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 "call/rtp_transport_config.h"
 #include "rtc_base/checks.h"
-#include "system_wrappers/include/field_trial.h"
+#include "rtc_base/experiments/field_trial_list.h"
+#include "rtc_base/experiments/field_trial_parser.h"
 
 namespace webrtc {
 namespace {
-bool ParseConfigParam(std::string exp_name, int* field) {
-  std::string group = field_trial::FindFullName(exp_name);
+using TimeScopedNetworkConfig = DegradedCall::TimeScopedNetworkConfig;
+
+bool ParseConfigParam(const WebRtcKeyValueConfig& trials,
+                      absl::string_view exp_name,
+                      int* field) {
+  std::string group = trials.Lookup(exp_name);
   if (group.empty())
     return false;
 
   return (sscanf(group.c_str(), "%d", field) == 1);
 }
 
-absl::optional<webrtc::BuiltInNetworkBehaviorConfig> ParseDegradationConfig(
+absl::optional<TimeScopedNetworkConfig> ParseDegradationConfig(
+    const WebRtcKeyValueConfig& trials,
     bool send) {
   std::string exp_prefix = "WebRTCFakeNetwork";
   if (send) {
@@ -43,33 +51,92 @@ absl::optional<webrtc::BuiltInNetworkBehaviorConfig> ParseDegradationConfig(
     exp_prefix += "Receive";
   }
 
-  webrtc::BuiltInNetworkBehaviorConfig config;
+  TimeScopedNetworkConfig config;
   bool configured = false;
   configured |=
-      ParseConfigParam(exp_prefix + "DelayMs", &config.queue_delay_ms);
-  configured |= ParseConfigParam(exp_prefix + "DelayStdDevMs",
+      ParseConfigParam(trials, exp_prefix + "DelayMs", &config.queue_delay_ms);
+  configured |= ParseConfigParam(trials, exp_prefix + "DelayStdDevMs",
                                  &config.delay_standard_deviation_ms);
   int queue_length = 0;
-  if (ParseConfigParam(exp_prefix + "QueueLength", &queue_length)) {
+  if (ParseConfigParam(trials, 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);
+  configured |= ParseConfigParam(trials, exp_prefix + "CapacityKbps",
+                                 &config.link_capacity_kbps);
+  configured |= ParseConfigParam(trials, exp_prefix + "LossPercent",
+                                 &config.loss_percent);
   int allow_reordering = 0;
-  if (ParseConfigParam(exp_prefix + "AllowReordering", &allow_reordering)) {
+  if (ParseConfigParam(trials, exp_prefix + "AllowReordering",
+                       &allow_reordering)) {
     config.allow_reordering = true;
     configured = true;
   }
-  configured |= ParseConfigParam(exp_prefix + "AvgBurstLossLength",
+  configured |= ParseConfigParam(trials, exp_prefix + "AvgBurstLossLength",
                                  &config.avg_burst_loss_length);
-  return configured
-             ? absl::optional<webrtc::BuiltInNetworkBehaviorConfig>(config)
+  return configured ? absl::optional<TimeScopedNetworkConfig>(config)
                     : absl::nullopt;
 }
+
+std::vector<TimeScopedNetworkConfig> GetNetworkConfigs(
+    const WebRtcKeyValueConfig& 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("codel_active_queue_management",
+                              [](TimeScopedNetworkConfig* p) {
+                                return &p->codel_active_queue_management;
+                              }),
+       FieldTrialStructMember(
+           "duration",
+           [](TimeScopedNetworkConfig* p) { return &p->duration; })},
+      {});
+  ParseFieldTrial({&trials_list},
+                  trials.Lookup(send ? "WebRTC-FakeNetworkSendConfig"
+                                     : "WebRTC-FakeNetworkReceiveConfig"));
+  std::vector<TimeScopedNetworkConfig> configs = trials_list.Get();
+  if (configs.empty()) {
+    // Try legacy fallback trials.
+    absl::optional<DegradedCall::TimeScopedNetworkConfig> fallback_config =
+        ParseDegradationConfig(trials, send);
+    if (fallback_config.has_value()) {
+      configs.push_back(*fallback_config);
+    }
+  }
+  return configs;
+}
+
 }  // namespace
 
 CallFactory::CallFactory() {
@@ -78,14 +145,18 @@ CallFactory::CallFactory() {
 
 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);
+  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();
 
-  if (send_degradation_config || receive_degradation_config) {
+  if (!send_degradation_configs.empty() ||
+      !receive_degradation_configs.empty()) {
     return new DegradedCall(
         std::unique_ptr<Call>(Call::Create(
             config, Clock::GetRealTimeClock(),
@@ -94,7 +165,7 @@ Call* CallFactory::CreateCall(const Call::Config& config) {
             config.rtp_transport_controller_send_factory->Create(
                 transportConfig, Clock::GetRealTimeClock(),
                 ProcessThread::Create("PacerThread")))),
-        send_degradation_config, receive_degradation_config,
+        send_degradation_configs, receive_degradation_configs,
         config.task_queue_factory);
   }
 

call/degraded_call.cc

@@ -127,27 +127,39 @@ bool DegradedCall::FakeNetworkPipeTransportAdapter::SendRtcp(
 
 DegradedCall::DegradedCall(
     std::unique_ptr<Call> call,
-    absl::optional<BuiltInNetworkBehaviorConfig> send_config,
-    absl::optional<BuiltInNetworkBehaviorConfig> receive_config,
+    const std::vector<TimeScopedNetworkConfig>& send_configs,
+    const std::vector<TimeScopedNetworkConfig>& receive_configs,
     TaskQueueFactory* task_queue_factory)
     : clock_(Clock::GetRealTimeClock()),
       call_(std::move(call)),
       task_queue_factory_(task_queue_factory),
-      send_config_(send_config),
+      send_config_index_(0),
+      send_configs_(send_configs),
       send_simulated_network_(nullptr),
-      receive_config_(receive_config) {
-  if (receive_config_) {
-    auto network = std::make_unique<SimulatedNetwork>(*receive_config_);
+      receive_config_index_(0),
+      receive_configs_(receive_configs) {
+  if (!receive_configs_.empty()) {
+    auto network = std::make_unique<SimulatedNetwork>(receive_configs_[0]);
     receive_simulated_network_ = network.get();
     receive_pipe_ =
         std::make_unique<webrtc::FakeNetworkPipe>(clock_, std::move(network));
     receive_pipe_->SetReceiver(call_->Receiver());
+    if (receive_configs_.size() > 1) {
+      call_->network_thread()->PostDelayedTask(
+          ToQueuedTask(task_safety_, [this] { UpdateReceiveNetworkConfig(); }),
+          receive_configs_[0].duration.ms());
     }
-  if (send_config_) {
-    auto network = std::make_unique<SimulatedNetwork>(*send_config_);
+  }
+  if (!send_configs_.empty()) {
+    auto network = std::make_unique<SimulatedNetwork>(send_configs_[0]);
     send_simulated_network_ = network.get();
     send_pipe_ = std::make_unique<FakeNetworkPipeOnTaskQueue>(
         task_queue_factory_, clock_, std::move(network));
+    if (send_configs_.size() > 1) {
+      call_->network_thread()->PostDelayedTask(
+          ToQueuedTask(task_safety_, [this] { UpdateSendNetworkConfig(); }),
+          send_configs_[0].duration.ms());
+    }
   }
 }
 
@@ -155,7 +167,7 @@ DegradedCall::~DegradedCall() = default;
 
 AudioSendStream* DegradedCall::CreateAudioSendStream(
     const AudioSendStream::Config& config) {
-  if (send_config_) {
+  if (!send_configs_.empty()) {
     auto transport_adapter = std::make_unique<FakeNetworkPipeTransportAdapter>(
         send_pipe_.get(), call_.get(), clock_, config.send_transport);
     AudioSendStream::Config degrade_config = config;
@@ -189,7 +201,7 @@ VideoSendStream* DegradedCall::CreateVideoSendStream(
     VideoSendStream::Config config,
     VideoEncoderConfig encoder_config) {
   std::unique_ptr<FakeNetworkPipeTransportAdapter> transport_adapter;
-  if (send_config_) {
+  if (!send_configs_.empty()) {
     transport_adapter = std::make_unique<FakeNetworkPipeTransportAdapter>(
         send_pipe_.get(), call_.get(), clock_, config.send_transport);
     config.send_transport = transport_adapter.get();
@@ -207,7 +219,7 @@ VideoSendStream* DegradedCall::CreateVideoSendStream(
     VideoEncoderConfig encoder_config,
     std::unique_ptr<FecController> fec_controller) {
   std::unique_ptr<FakeNetworkPipeTransportAdapter> transport_adapter;
-  if (send_config_) {
+  if (!send_configs_.empty()) {
     transport_adapter = std::make_unique<FakeNetworkPipeTransportAdapter>(
         send_pipe_.get(), call_.get(), clock_, config.send_transport);
     config.send_transport = transport_adapter.get();
@@ -251,7 +263,7 @@ void DegradedCall::AddAdaptationResource(
 }
 
 PacketReceiver* DegradedCall::Receiver() {
-  if (receive_config_) {
+  if (!receive_configs_.empty()) {
     return this;
   }
   return call_->Receiver();
@@ -299,7 +311,7 @@ void DegradedCall::OnUpdateSyncGroup(AudioReceiveStream& stream,
 }
 
 void DegradedCall::OnSentPacket(const rtc::SentPacket& sent_packet) {
-  if (send_config_) {
+  if (!send_configs_.empty()) {
     // If we have a degraded send-transport, we have already notified call
     // about the supposed network send time. Discard the actual network send
     // time in order to properly fool the BWE.
@@ -325,4 +337,21 @@ PacketReceiver::DeliveryStatus DegradedCall::DeliverPacket(
   receive_pipe_->Process();
   return status;
 }
+
+void DegradedCall::UpdateSendNetworkConfig() {
+  send_config_index_ = (send_config_index_ + 1) % send_configs_.size();
+  send_simulated_network_->SetConfig(send_configs_[send_config_index_]);
+  call_->network_thread()->PostDelayedTask(
+      ToQueuedTask(task_safety_, [this] { UpdateSendNetworkConfig(); }),
+      send_configs_[send_config_index_].duration.ms());
+}
+
+void DegradedCall::UpdateReceiveNetworkConfig() {
+  receive_config_index_ = (receive_config_index_ + 1) % receive_configs_.size();
+  receive_simulated_network_->SetConfig(
+      receive_configs_[receive_config_index_]);
+  call_->network_thread()->PostDelayedTask(
+      ToQueuedTask(task_safety_, [this] { UpdateReceiveNetworkConfig(); }),
+      receive_configs_[receive_config_index_].duration.ms());
+}
 }  // namespace webrtc

call/degraded_call.h

@@ -17,6 +17,7 @@
 #include <map>
 #include <memory>
 #include <string>
+#include <vector>
 
 #include "absl/types/optional.h"
 #include "api/call/transport.h"
@@ -38,15 +39,20 @@
 #include "rtc_base/copy_on_write_buffer.h"
 #include "rtc_base/network/sent_packet.h"
 #include "rtc_base/task_queue.h"
+#include "rtc_base/task_utils/pending_task_safety_flag.h"
 #include "system_wrappers/include/clock.h"
 
 namespace webrtc {
 class DegradedCall : public Call, private PacketReceiver {
  public:
+  struct TimeScopedNetworkConfig : public BuiltInNetworkBehaviorConfig {
+    TimeDelta duration = TimeDelta::PlusInfinity();
+  };
+
   explicit DegradedCall(
       std::unique_ptr<Call> call,
-      absl::optional<BuiltInNetworkBehaviorConfig> send_config,
-      absl::optional<BuiltInNetworkBehaviorConfig> receive_config,
+      const std::vector<TimeScopedNetworkConfig>& send_configs,
+      const std::vector<TimeScopedNetworkConfig>& receive_configs,
       TaskQueueFactory* task_queue_factory);
   ~DegradedCall() override;
 
@@ -157,14 +163,17 @@ class DegradedCall : public Call, private PacketReceiver {
     Transport* const real_transport_;
   };
 
-  Clock* const clock_;
-  const std::unique_ptr<Call> call_;
-  TaskQueueFactory* const task_queue_factory_;
-
   void SetClientBitratePreferences(
       const webrtc::BitrateSettings& preferences) override {}
+  void UpdateSendNetworkConfig();
+  void UpdateReceiveNetworkConfig();
 
-  const absl::optional<BuiltInNetworkBehaviorConfig> send_config_;
+  Clock* const clock_;
+  const std::unique_ptr<Call> call_;
+  ScopedTaskSafety task_safety_;
+  TaskQueueFactory* const task_queue_factory_;
+  size_t send_config_index_;
+  const std::vector<TimeScopedNetworkConfig> send_configs_;
   SimulatedNetwork* send_simulated_network_;
   std::unique_ptr<FakeNetworkPipeOnTaskQueue> send_pipe_;
   std::map<AudioSendStream*, std::unique_ptr<FakeNetworkPipeTransportAdapter>>
@@ -172,7 +181,8 @@ class DegradedCall : public Call, private PacketReceiver {
   std::map<VideoSendStream*, std::unique_ptr<FakeNetworkPipeTransportAdapter>>
       video_send_transport_adapters_;
 
-  const absl::optional<BuiltInNetworkBehaviorConfig> receive_config_;
+  size_t receive_config_index_;
+  const std::vector<TimeScopedNetworkConfig> receive_configs_;
   SimulatedNetwork* receive_simulated_network_;
   std::unique_ptr<FakeNetworkPipe> receive_pipe_;
 };