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Unified Diff: webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter_unittest.cc

Issue 1329083005: Add TransportFeedback adapter, adapting remote feedback to bwe estiamtor (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years, 3 months ago
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Index: webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter_unittest.cc
diff --git a/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter_unittest.cc b/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter_unittest.cc
new file mode 100644
index 0000000000000000000000000000000000000000..5d63b899a35ea0069fcda55cb7fe9deb9e4c42ba
--- /dev/null
+++ b/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter_unittest.cc
@@ -0,0 +1,222 @@
+/*
+ * 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 understa
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
sprang_webrtc 2015/09/11 12:33:48 Yes, the instance will be owned by TransportFeedba
+ 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
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
sprang_webrtc 2015/09/11 12:33:48 Done.
+} // namespace test
+} // namespace webrtc

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