Add TransportFeedback adapter, adapting remote feedback to bwe estiamtor

webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter_unittest.cc

+/*
+ *  Copyright (c) 2015 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 <vector>
+
+#include "testing/gmock/include/gmock/gmock.h"
+#include "testing/gtest/include/gtest/gtest.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_estimator.h"
+#include "webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
+#include "webrtc/modules/utility/interface/mock/mock_process_thread.h"
+#include "webrtc/system_wrappers/interface/clock.h"
+
+using ::testing::_;
+using ::testing::Invoke;
+
+namespace webrtc {
+namespace test {
+
+class TransportFeedbackAdapterTest : public ::testing::Test {
+ public:
+  TransportFeedbackAdapterTest()
+      : clock_(0),
+        bitrate_estimator_(nullptr),
+        receiver_estimated_bitrate_(0) {}
+
+  virtual ~TransportFeedbackAdapterTest() {}
+
+  virtual void SetUp() {
+    adapter_.reset(new TransportFeedbackAdapter(
+        new RtcpBandwidthObserverAdapter(this), &clock_, &process_thread_));
+
+    bitrate_estimator_ = new MockRemoteBitrateEstimator();
+    EXPECT_CALL(process_thread_, RegisterModule(bitrate_estimator_)).Times(1);
+    adapter_->SetBitrateEstimator(bitrate_estimator_);
+  }
+
+  virtual void TearDown() {
+    EXPECT_CALL(process_thread_, DeRegisterModule(bitrate_estimator_)).Times(1);
+    adapter_.reset();
+  }
+
+ protected:
+  // Proxy class used since TransportFeedbackAdapter will own the instance
+  // passed at construction.
+  class RtcpBandwidthObserverAdapter : public RtcpBandwidthObserver {
+   public:
+    explicit RtcpBandwidthObserverAdapter(TransportFeedbackAdapterTest* owner)
+        : owner_(owner) {}
+
+    void OnReceivedEstimatedBitrate(uint32_t bitrate) override {
+      owner_->receiver_estimated_bitrate_ = bitrate;
+    }
+
+    void OnReceivedRtcpReceiverReport(const ReportBlockList& report_blocks,
+                                      int64_t rtt,
+                                      int64_t now_ms) override {
+      RTC_NOTREACHED();
+    }
+
+    TransportFeedbackAdapterTest* const owner_;
+  };
+
+  void OnReceivedEstimatedBitrate(uint32_t bitrate) {}
+
+  void OnReceivedRtcpReceiverReport(const ReportBlockList& report_blocks,
+                                    int64_t rtt,
+                                    int64_t now_ms) {}
+
+  void ComparePacketVectors(const std::vector<PacketInfo>& truth,
+                            const std::vector<PacketInfo>& input) {
+    ASSERT_EQ(truth.size(), input.size());
+    size_t len = truth.size();
+    int64_t arrival_time_delta =
+        truth[0].arrival_time_ms - input[0].arrival_time_ms;

[stefan-webrtc, 2015/09/11 08:15:12]

Just from reading this code I don't fully understand what this method is
checking. Naively, I would expect the delta here to be zero if things are
working well. Then you are adding this delta to input[i] and comparing it to
truth[i] at lines 88-89. Could you explain with a comment maybe?

[sprang_webrtc, 2015/09/11 12:33:47]

Done.

+    for (size_t i = 0; i < len; ++i) {
+      EXPECT_EQ(truth[i].arrival_time_ms,
+                input[i].arrival_time_ms + arrival_time_delta);
+      EXPECT_EQ(truth[i].send_time_ms, input[i].send_time_ms);
+      EXPECT_EQ(truth[i].sequence_number, input[i].sequence_number);
+      EXPECT_EQ(truth[i].payload_size, input[i].payload_size);
+      EXPECT_EQ(truth[i].was_paced, input[i].was_paced);
+    }
+  }
+
+  SimulatedClock clock_;
+  MockProcessThread process_thread_;
+  MockRemoteBitrateEstimator* bitrate_estimator_;

[stefan-webrtc, 2015/09/11 08:15:12]

Do you have to allocate it on the heap? Would save you a lot of * in front of
bitrate_estimator_ in the tests.

[sprang_webrtc, 2015/09/11 12:33:48]

Yes, the instance will be owned by TransportFeedbackAdapter...

+  rtc::scoped_ptr<TransportFeedbackAdapter> adapter_;
+
+  uint32_t receiver_estimated_bitrate_;
+};
+
+TEST_F(TransportFeedbackAdapterTest, AdaptsFeedbackAndPopulatesSendTimes) {
+  std::vector<PacketInfo> packets;
+  packets.push_back(PacketInfo(100, 200, 0, 1500, true));
+  packets.push_back(PacketInfo(110, 210, 1, 1500, true));
+  packets.push_back(PacketInfo(120, 220, 2, 1500, true));
+  packets.push_back(PacketInfo(130, 230, 3, 1500, true));
+  packets.push_back(PacketInfo(140, 240, 4, 1500, true));
+
+  for (const PacketInfo& packet : packets) {
+    adapter_->OnPacketSent(packet.sequence_number, packet.send_time_ms,
+                           packet.payload_size, packet.was_paced);

[stefan-webrtc, 2015/09/11 08:15:12]

Makes me wonder why we don't pass in a PacketInfo object into this method :)

[sprang_webrtc, 2015/09/11 12:33:47]

Done.

+  }
+
+  rtcp::TransportFeedback feedback;
+  feedback.WithBase(packets[0].sequence_number,
+                    packets[0].arrival_time_ms * 1000);
+
+  for (const PacketInfo& packet : packets) {
+    EXPECT_TRUE(feedback.WithReceivedPacket(packet.sequence_number,
+                                            packet.arrival_time_ms * 1000));
+  }
+
+  feedback.Build();
+
+  EXPECT_CALL(*bitrate_estimator_, IncomingPacketFeedbackVector(_))
+      .Times(1)
+      .WillOnce(Invoke(
+          [packets, this](const std::vector<PacketInfo>& feedback_vector) {
+            ComparePacketVectors(packets, feedback_vector);
+          }));
+  adapter_->OnTransportFeedback(feedback);
+}
+
+TEST_F(TransportFeedbackAdapterTest, HandlesDroppedPackets) {
+  std::vector<PacketInfo> packets;
+  packets.push_back(PacketInfo(100, 200, 0, 1500, true));
+  packets.push_back(PacketInfo(110, 210, 1, 1500, true));
+  packets.push_back(PacketInfo(120, 220, 2, 1500, true));
+  packets.push_back(PacketInfo(130, 230, 3, 1500, true));
+  packets.push_back(PacketInfo(140, 240, 4, 1500, true));
+
+  const uint16_t kSendSideDropBefore = 1;
+  const uint16_t kReceiveSideDropAfter = 3;
+
+  for (const PacketInfo& packet : packets) {
+    if (packet.sequence_number >= kSendSideDropBefore) {
+      adapter_->OnPacketSent(packet.sequence_number, packet.send_time_ms,
+                             packet.payload_size, packet.was_paced);
+    }
+  }
+
+  rtcp::TransportFeedback feedback;
+  feedback.WithBase(packets[0].sequence_number,
+                    packets[0].arrival_time_ms * 1000);
+
+  for (const PacketInfo& packet : packets) {
+    if (packet.sequence_number <= kReceiveSideDropAfter) {
+      EXPECT_TRUE(feedback.WithReceivedPacket(packet.sequence_number,
+                                              packet.arrival_time_ms * 1000));
+    }
+  }
+
+  feedback.Build();
+
+  std::vector<PacketInfo> expected_packets(
+      packets.begin() + kSendSideDropBefore,
+      packets.begin() + kReceiveSideDropAfter + 1);
+
+  EXPECT_CALL(*bitrate_estimator_, IncomingPacketFeedbackVector(_))
+      .Times(1)
+      .WillOnce(Invoke([expected_packets,
+                        this](const std::vector<PacketInfo>& feedback_vector) {
+        ComparePacketVectors(expected_packets, feedback_vector);
+      }));
+  adapter_->OnTransportFeedback(feedback);
+}
+
+TEST_F(TransportFeedbackAdapterTest, SendTimeWrapsBothWays) {
+  int64_t kHighArrivalTimeMs = rtcp::TransportFeedback::kDeltaScaleFactor *
+                               (1L << 8) * ((1L << 23) - 1) / 1000;
+  std::vector<PacketInfo> packets;
+  packets.push_back(PacketInfo(kHighArrivalTimeMs - 64, 200, 0, 1500, true));
+  packets.push_back(PacketInfo(kHighArrivalTimeMs + 64, 210, 1, 1500, true));
+  packets.push_back(PacketInfo(kHighArrivalTimeMs, 220, 2, 1500, true));
+
+  for (const PacketInfo& packet : packets) {
+    adapter_->OnPacketSent(packet.sequence_number, packet.send_time_ms,
+                           packet.payload_size, packet.was_paced);
+  }
+
+  for (size_t i = 0; i < packets.size(); ++i) {
+    rtc::scoped_ptr<rtcp::TransportFeedback> feedback(
+        new rtcp::TransportFeedback());
+    feedback->WithBase(packets[i].sequence_number,
+                       packets[i].arrival_time_ms * 1000);
+
+    EXPECT_TRUE(feedback->WithReceivedPacket(
+        packets[i].sequence_number, packets[i].arrival_time_ms * 1000));
+
+    rtc::scoped_ptr<rtcp::RawPacket> raw_packet = feedback->Build();
+    feedback = rtcp::TransportFeedback::ParseFrom(raw_packet->Buffer(),
+                                                  raw_packet->Length());
+
+    std::vector<PacketInfo> expected_packets;
+    expected_packets.push_back(packets[i]);
+
+    EXPECT_CALL(*bitrate_estimator_, IncomingPacketFeedbackVector(_))
+        .Times(1)
+        .WillOnce(Invoke([expected_packets, this](
+            const std::vector<PacketInfo>& feedback_vector) {
+          ComparePacketVectors(expected_packets, feedback_vector);
+        }));
+    adapter_->OnTransportFeedback(*feedback.get());
+  }
+}
+

[stefan-webrtc, 2015/09/11 08:15:12]

Should we have a test for the limit on time deltas as well?

[sprang_webrtc, 2015/09/11 12:33:48]

Done.

+}  // namespace test
+}  // namespace webrtc