Add NACK rate throttling for audio channels.

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.
  */
 
@@ skipping 993 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, nullptr, &send_packet_observer_, &retransmission_rate_limiter_));
   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, nullptr, &mock_rtc_event_log_, nullptr,
+      &retransmission_rate_limiter_));
 
   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));
@@ skipping 123 matching lines @@
       ++num_calls_;
       ssrc_ = ssrc;
       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));
+  rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport_,
+                                  &mock_paced_sender_, nullptr, nullptr,
+                                  nullptr, &callback, nullptr, nullptr, nullptr,
+                                  &retransmission_rate_limiter_));
 
   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 38 matching lines @@
       ssrc_ = ssrc;
       total_bitrate_ = total_bitrate;
       retransmit_bitrate_ = retransmit_bitrate;
     }
 
     uint32_t num_calls_;
     uint32_t ssrc_;
     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));
+  rtp_sender_.reset(new RTPSender(
+      false, &fake_clock_, &transport_, nullptr, nullptr, nullptr, &callback,
+      nullptr, nullptr, nullptr, nullptr, &retransmission_rate_limiter_));
 
   // 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;
@@ skipping 34 matching lines @@
 
   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));
+    rtp_sender_.reset(new RTPSender(
+        true, &fake_clock_, &transport_, nullptr, nullptr, nullptr, nullptr,
+        nullptr, nullptr, nullptr, nullptr, &retransmission_rate_limiter_));
     rtp_sender_->SetSequenceNumber(kSeqNum);
   }
 };
 
 TEST_F(RtpSenderTestWithoutPacer, StreamDataCountersCallbacks) {
   class TestCallback : public StreamDataCountersCallback {
    public:
     TestCallback() : StreamDataCountersCallback(), ssrc_(0), counters_() {}
     virtual ~TestCallback() {}
 
@@ skipping 307 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