Refactor NACK bitrate allocation

webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc

 /*
  *  Copyright (c) 2012 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 <list>
 #include <memory>
 #include <vector>
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/buffer.h"
+#include "webrtc/base/rate_limiter.h"
 #include "webrtc/call/mock/mock_rtc_event_log.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/system_wrappers/include/stl_util.h"
@@ skipping 98 matching lines @@
  public:
   MOCK_METHOD3(OnSendPacket, void(uint16_t, int64_t, uint32_t));
 };
 
 class RtpSenderTest : public ::testing::Test {
  protected:
   RtpSenderTest()
       : fake_clock_(kStartTime),
         mock_rtc_event_log_(),
         mock_paced_sender_(),
+        retransmission_rate_limiter_(&fake_clock_, 1000),
         rtp_sender_(),
         payload_(kPayload),
         transport_(),
-        kMarkerBit(true) {
-  }
+        kMarkerBit(true) {}
 
   void SetUp() override { SetUpRtpSender(true); }
 
   void SetUpRtpSender(bool pacer) {
     rtp_sender_.reset(new RTPSender(
         false, &fake_clock_, &transport_, pacer ? &mock_paced_sender_ : nullptr,
         &seq_num_allocator_, nullptr, nullptr, nullptr, nullptr,
-        &mock_rtc_event_log_, &send_packet_observer_));
+        &mock_rtc_event_log_, &send_packet_observer_,
+        &retransmission_rate_limiter_));
     rtp_sender_->SetSequenceNumber(kSeqNum);
   }
 
   SimulatedClock fake_clock_;
   MockRtcEventLog mock_rtc_event_log_;
   MockRtpPacketSender mock_paced_sender_;
   MockTransportSequenceNumberAllocator seq_num_allocator_;
   MockSendPacketObserver send_packet_observer_;
+  RateLimiter retransmission_rate_limiter_;
   std::unique_ptr<RTPSender> rtp_sender_;
   int payload_;
   LoopbackTransportTest transport_;
   const bool kMarkerBit;
   uint8_t packet_[kMaxPacketLength];
 
   void VerifyRTPHeaderCommon(const RTPHeader& rtp_header) {
     VerifyRTPHeaderCommon(rtp_header, kMarkerBit, 0);
   }
 
@@ skipping 569 matching lines @@
   EXPECT_CALL(mock_rtc_event_log_,
               LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _));
 
   rtp_sender_->SetStorePacketsStatus(true, 10);
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
                    kRtpExtensionTransmissionTimeOffset,
                    kTransmissionTimeOffsetExtensionId));
   EXPECT_EQ(
       0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
                                                  kAbsoluteSendTimeExtensionId));
-  rtp_sender_->SetTargetBitrate(300000);
   int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
   int rtp_length_int = rtp_sender_->BuildRTPheader(
       packet_, kPayload, kMarkerBit, kTimestamp, capture_time_ms);
   ASSERT_NE(-1, rtp_length_int);
   size_t rtp_length = static_cast<size_t>(rtp_length_int);
 
   // Packet should be stored in a send bucket.
   EXPECT_EQ(0, rtp_sender_->SendToNetwork(packet_, 0, rtp_length,
                                           capture_time_ms, kAllowRetransmission,
                                           RtpPacketSender::kNormalPriority));
@@ skipping 33 matching lines @@
   EXPECT_CALL(mock_rtc_event_log_,
               LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _));
 
   rtp_sender_->SetStorePacketsStatus(true, 10);
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
                    kRtpExtensionTransmissionTimeOffset,
                    kTransmissionTimeOffsetExtensionId));
   EXPECT_EQ(
       0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
                                                  kAbsoluteSendTimeExtensionId));
-  rtp_sender_->SetTargetBitrate(300000);
   int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
   int rtp_length_int = rtp_sender_->BuildRTPheader(
       packet_, kPayload, kMarkerBit, kTimestamp, capture_time_ms);
   ASSERT_NE(-1, rtp_length_int);
   size_t rtp_length = static_cast<size_t>(rtp_length_int);
 
   // Packet should be stored in a send bucket.
   EXPECT_EQ(0, rtp_sender_->SendToNetwork(packet_, 0, rtp_length,
                                           capture_time_ms, kAllowRetransmission,
                                           RtpPacketSender::kNormalPriority));
@@ skipping 61 matching lines @@
   // Create and set up parser.
   std::unique_ptr<webrtc::RtpHeaderParser> rtp_parser(
       webrtc::RtpHeaderParser::Create());
   ASSERT_TRUE(rtp_parser.get() != nullptr);
   rtp_parser->RegisterRtpHeaderExtension(kRtpExtensionTransmissionTimeOffset,
                                          kTransmissionTimeOffsetExtensionId);
   rtp_parser->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
                                          kAbsoluteSendTimeExtensionId);
   webrtc::RTPHeader rtp_header;
 
-  rtp_sender_->SetTargetBitrate(300000);
   int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
   int rtp_length_int = rtp_sender_->BuildRTPheader(
       packet_, kPayload, kMarkerBit, timestamp, capture_time_ms);
   const uint32_t media_packet_timestamp = timestamp;
   ASSERT_NE(-1, rtp_length_int);
   size_t rtp_length = static_cast<size_t>(rtp_length_int);
 
   // Packet should be stored in a send bucket.
   EXPECT_EQ(0, rtp_sender_->SendToNetwork(packet_, 0, rtp_length,
                                           capture_time_ms, kAllowRetransmission,
@@ skipping 111 matching lines @@
   const bool kIsRetransmit = true;
   rtp_sender_->TimeToSendPacket(kSeqNum, fake_clock_.TimeInMilliseconds(),
                                 kIsRetransmit, PacketInfo::kNotAProbe);
   EXPECT_EQ(1, transport_.packets_sent_);
 }
 
 TEST_F(RtpSenderTest, OnSendPacketNotUpdatedWithoutSeqNumAllocator) {
   rtp_sender_.reset(new RTPSender(
       false, &fake_clock_, &transport_, &mock_paced_sender_,
       nullptr /* TransportSequenceNumberAllocator */, nullptr, nullptr, nullptr,
-      nullptr, nullptr, &send_packet_observer_));
+      nullptr, nullptr, &send_packet_observer_, nullptr));
   rtp_sender_->SetSequenceNumber(kSeqNum);
   rtp_sender_->SetStorePacketsStatus(true, 10);
 
   EXPECT_CALL(send_packet_observer_, OnSendPacket(_, _, _)).Times(0);
   EXPECT_CALL(mock_paced_sender_, InsertPacket(_, _, _, _, _, _)).Times(1);
 
   SendGenericPayload();  // Packet passed to pacer.
   const bool kIsRetransmit = false;
   rtp_sender_->TimeToSendPacket(kSeqNum, fake_clock_.TimeInMilliseconds(),
                                 kIsRetransmit, PacketInfo::kNotAProbe);
   EXPECT_EQ(1, transport_.packets_sent_);
 }
 
 TEST_F(RtpSenderTest, SendRedundantPayloads) {
   MockTransport transport;
   rtp_sender_.reset(new RTPSender(
       false, &fake_clock_, &transport, &mock_paced_sender_, nullptr, nullptr,
-      nullptr, nullptr, nullptr, &mock_rtc_event_log_, nullptr));
+      nullptr, nullptr, nullptr, &mock_rtc_event_log_, nullptr, nullptr));
 
   rtp_sender_->SetSequenceNumber(kSeqNum);
   rtp_sender_->SetRtxPayloadType(kRtxPayload, kPayload);
 
   uint16_t seq_num = kSeqNum;
   rtp_sender_->SetStorePacketsStatus(true, 10);
   int32_t rtp_header_len = kRtpHeaderSize;
   EXPECT_EQ(
       0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
                                                  kAbsoluteSendTimeExtensionId));
   rtp_header_len += 4;  // 4 bytes extension.
   rtp_header_len += 4;  // 4 extra bytes common to all extension headers.
 
   rtp_sender_->SetRtxStatus(kRtxRetransmitted | kRtxRedundantPayloads);
   rtp_sender_->SetRtxSsrc(1234);
 
   // Create and set up parser.
   std::unique_ptr<webrtc::RtpHeaderParser> rtp_parser(
       webrtc::RtpHeaderParser::Create());
   ASSERT_TRUE(rtp_parser.get() != nullptr);
   rtp_parser->RegisterRtpHeaderExtension(kRtpExtensionTransmissionTimeOffset,
                                          kTransmissionTimeOffsetExtensionId);
   rtp_parser->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
                                          kAbsoluteSendTimeExtensionId);
-  rtp_sender_->SetTargetBitrate(300000);
   const size_t kNumPayloadSizes = 10;
   const size_t kPayloadSizes[kNumPayloadSizes] = {500, 550, 600, 650, 700,
                                                   750, 800, 850, 900, 950};
   // Expect all packets go through the pacer.
   EXPECT_CALL(mock_paced_sender_,
               InsertPacket(RtpPacketSender::kNormalPriority, _, _, _, _, _))
       .Times(kNumPayloadSizes);
   EXPECT_CALL(mock_rtc_event_log_,
               LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _))
       .Times(kNumPayloadSizes);
@@ skipping 101 matching lines @@
       frame_counts_ = frame_counts;
     }
 
     uint32_t num_calls_;
     uint32_t ssrc_;
     FrameCounts frame_counts_;
   } callback;
 
   rtp_sender_.reset(new RTPSender(
       false, &fake_clock_, &transport_, &mock_paced_sender_, nullptr, nullptr,
-      nullptr, &callback, nullptr, nullptr, nullptr));
+      nullptr, &callback, nullptr, nullptr, nullptr, nullptr));
 
   char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC";
   const uint8_t payload_type = 127;
   ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 90000,
                                             0, 1500));
   uint8_t payload[] = {47, 11, 32, 93, 89};
   rtp_sender_->SetStorePacketsStatus(true, 1);
   uint32_t ssrc = rtp_sender_->SSRC();
 
   EXPECT_CALL(mock_paced_sender_, InsertPacket(_, _, _, _, _, _))
@@ skipping 16 matching lines @@
   EXPECT_EQ(ssrc, callback.ssrc_);
   EXPECT_EQ(1, callback.frame_counts_.key_frames);
   EXPECT_EQ(1, callback.frame_counts_.delta_frames);
 
   rtp_sender_.reset();
 }
 
 TEST_F(RtpSenderTest, BitrateCallbacks) {
   class TestCallback : public BitrateStatisticsObserver {
    public:
-    TestCallback() : BitrateStatisticsObserver(), num_calls_(0), ssrc_(0) {}
+    TestCallback()
+        : BitrateStatisticsObserver(),
+          num_calls_(0),
+          ssrc_(0),
+          total_bitrate_(0),
+          retransmit_bitrate_(0) {}
     virtual ~TestCallback() {}
 
-    void Notify(const BitrateStatistics& total_stats,
-                const BitrateStatistics& retransmit_stats,
+    void Notify(uint32_t total_bitrate,
+                uint32_t retransmit_bitrate,
                 uint32_t ssrc) override {
       ++num_calls_;
       ssrc_ = ssrc;
-      total_stats_ = total_stats;
-      retransmit_stats_ = retransmit_stats;
+      total_bitrate_ = total_bitrate;
+      retransmit_bitrate_ = retransmit_bitrate;
     }
 
     uint32_t num_calls_;
     uint32_t ssrc_;
-    BitrateStatistics total_stats_;
-    BitrateStatistics retransmit_stats_;
+    uint32_t total_bitrate_;
+    uint32_t retransmit_bitrate_;
   } callback;
   rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport_, nullptr,
                                   nullptr, nullptr, &callback, nullptr, nullptr,
-                                  nullptr, nullptr));
+                                  nullptr, nullptr, nullptr));
 
-  // Simulate kNumPackets sent with kPacketInterval ms intervals.
-  const uint32_t kNumPackets = 15;
+  // Simulate kNumPackets sent with kPacketInterval ms intervals, with the
+  // number of packets selected so that we fill (but don't overflow) the one
+  // second averaging window.
+  const uint32_t kWindowSizeMs = 1000;
   const uint32_t kPacketInterval = 20;
+  const uint32_t kNumPackets =
+      (kWindowSizeMs - kPacketInterval) / kPacketInterval;
   // Overhead = 12 bytes RTP header + 1 byte generic header.
   const uint32_t kPacketOverhead = 13;
 
   char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC";
   const uint8_t payload_type = 127;
   ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 90000,
                                             0, 1500));
   uint8_t payload[] = {47, 11, 32, 93, 89};
   rtp_sender_->SetStorePacketsStatus(true, 1);
   uint32_t ssrc = rtp_sender_->SSRC();
 
   // Initial process call so we get a new time window.
   rtp_sender_->ProcessBitrate();
-  uint64_t start_time = fake_clock_.CurrentNtpInMilliseconds();
 
   // Send a few frames.
   for (uint32_t i = 0; i < kNumPackets; ++i) {
     ASSERT_EQ(0,
               rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234,
                                             4321, payload, sizeof(payload), 0));
     fake_clock_.AdvanceTimeMilliseconds(kPacketInterval);
   }
 
   rtp_sender_->ProcessBitrate();
 
-  const uint32_t expected_packet_rate = 1000 / kPacketInterval;
-
   // We get one call for every stats updated, thus two calls since both the
   // stream stats and the retransmit stats are updated once.
   EXPECT_EQ(2u, callback.num_calls_);
   EXPECT_EQ(ssrc, callback.ssrc_);
-  EXPECT_EQ(start_time + (kNumPackets * kPacketInterval),
-            callback.total_stats_.timestamp_ms);
-  EXPECT_EQ(expected_packet_rate, callback.total_stats_.packet_rate);
-  EXPECT_EQ((kPacketOverhead + sizeof(payload)) * 8 * expected_packet_rate,
-            callback.total_stats_.bitrate_bps);
+  const uint32_t kTotalPacketSize = kPacketOverhead + sizeof(payload);
+  // Bitrate measured over delta between last and first timestamp, plus one.
+  const uint32_t kExpectedWindowMs = kNumPackets * kPacketInterval + 1;
+  const uint32_t kExpectedBitsAccumulated = kTotalPacketSize * kNumPackets * 8;
+  const uint32_t kExpectedRateBps =
+      (kExpectedBitsAccumulated * 1000 + (kExpectedWindowMs / 2)) /
+      kExpectedWindowMs;
+  EXPECT_EQ(kExpectedRateBps, callback.total_bitrate_);
 
   rtp_sender_.reset();
 }
 
 class RtpSenderAudioTest : public RtpSenderTest {
  protected:
   RtpSenderAudioTest() {}
 
   void SetUp() override {
     payload_ = kAudioPayload;
     rtp_sender_.reset(new RTPSender(true, &fake_clock_, &transport_, nullptr,
                                     nullptr, nullptr, nullptr, nullptr, nullptr,
-                                    nullptr, nullptr));
+                                    nullptr, nullptr, nullptr));
     rtp_sender_->SetSequenceNumber(kSeqNum);
   }
 };
 
 TEST_F(RtpSenderTestWithoutPacer, StreamDataCountersCallbacks) {
   class TestCallback : public StreamDataCountersCallback {
    public:
     TestCallback() : StreamDataCountersCallback(), ssrc_(0), counters_() {}
     virtual ~TestCallback() {}
 
@@ skipping 247 matching lines @@
 
   EXPECT_EQ(
       transport_.total_bytes_sent_,
       rtp_stats.transmitted.TotalBytes() + rtx_stats.transmitted.TotalBytes());
 }
 
 TEST_F(RtpSenderTestWithoutPacer, RespectsNackBitrateLimit) {
   const int32_t kPacketSize = 1400;
   const int32_t kNumPackets = 30;
 
+  retransmission_rate_limiter_.SetMaxRate(kPacketSize * kNumPackets * 8);
+
   rtp_sender_->SetStorePacketsStatus(true, kNumPackets);
-  // Set bitrate (in kbps) to fit kNumPackets á kPacketSize bytes in one second.
-  rtp_sender_->SetTargetBitrate(kNumPackets * kPacketSize * 8);
   const uint16_t kStartSequenceNumber = rtp_sender_->SequenceNumber();
   std::list<uint16_t> sequence_numbers;
   for (int32_t i = 0; i < kNumPackets; ++i) {
     sequence_numbers.push_back(kStartSequenceNumber + i);
     fake_clock_.AdvanceTimeMilliseconds(1);
     SendPacket(fake_clock_.TimeInMilliseconds(), kPacketSize);
   }
   EXPECT_EQ(kNumPackets, transport_.packets_sent_);
 
   fake_clock_.AdvanceTimeMilliseconds(1000 - kNumPackets);
 
   // Resending should work - brings the bandwidth up to the limit.
   // NACK bitrate is capped to the same bitrate as the encoder, since the max
   // protection overhead is 50% (see MediaOptimization::SetTargetRates).
   rtp_sender_->OnReceivedNACK(sequence_numbers, 0);
   EXPECT_EQ(kNumPackets * 2, transport_.packets_sent_);
 
+  // Must be at least 5ms in between retransmission attempts.
+  fake_clock_.AdvanceTimeMilliseconds(5);
+
   // Resending should not work, bandwidth exceeded.
   rtp_sender_->OnReceivedNACK(sequence_numbers, 0);
   EXPECT_EQ(kNumPackets * 2, transport_.packets_sent_);
 }
 
 // Verify that all packets of a frame have CVO byte set.
 TEST_F(RtpSenderVideoTest, SendVideoWithCVO) {
   RTPVideoHeader hdr = {0};
   hdr.rotation = kVideoRotation_90;
 
@@ skipping 17 matching lines @@
       reinterpret_cast<uint8_t*>(transport_.sent_packets_[0]->data()),
       transport_.sent_packets_[0]->size(), true, &map, kSeqNum, hdr.rotation);
 
   // Verify that this packet does have CVO byte.
   VerifyCVOPacket(
       reinterpret_cast<uint8_t*>(transport_.sent_packets_[1]->data()),
       transport_.sent_packets_[1]->size(), true, &map, kSeqNum + 1,
       hdr.rotation);
 }
 }  // namespace webrtc