Refactor NACK bitrate allocation

webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index f0b6411af2711ffc3d2d39c446774268e5ecebec..dd5f0e0b22761b4aea34dbbcfa33e342746b016d 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -114,6 +114,9 @@ class MockRtpPacketSender : public RtpPacketSender {
                     int64_t capture_time_ms,
                     size_t bytes,
                     bool retransmission));
+
+  MOCK_METHOD1(AllocateRetransmissionBitrate, bool(size_t bytes));
+  MOCK_METHOD0(CurrentRetransmissionBitrate, int(void));
 };
 
 class MockTransportSequenceNumberAllocator
@@ -811,6 +814,8 @@ TEST_F(RtpSenderTest, TrafficSmoothingRetransmits) {
 
   EXPECT_EQ(0, transport_.packets_sent_);
 
+  EXPECT_CALL(mock_paced_sender_, AllocateRetransmissionBitrate(rtp_length))
+      .WillOnce(testing::Return(true));
   EXPECT_CALL(mock_paced_sender_, InsertPacket(RtpPacketSender::kNormalPriority,
                                                _, kSeqNum, _, _, _));
 
@@ -1213,30 +1218,39 @@ TEST_F(RtpSenderTest, FrameCountCallbacks) {
 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));
 
-  // 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;
 
@@ -1250,7 +1264,6 @@ TEST_F(RtpSenderTest, BitrateCallbacks) {
 
   // 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) {
@@ -1262,17 +1275,18 @@ TEST_F(RtpSenderTest, BitrateCallbacks) {
 
   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();
 }