Add unit test for nack bandwidth constraint.

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 1330 matching lines @@
   EXPECT_EQ(rtx_stats.transmitted.TotalBytes(),
       rtx_stats.transmitted.payload_bytes +
       rtx_stats.transmitted.header_bytes +
       rtx_stats.transmitted.padding_bytes);
 
   EXPECT_EQ(transport_.total_bytes_sent_,
             rtp_stats.transmitted.TotalBytes() +
             rtx_stats.transmitted.TotalBytes());
 }
 
+TEST_F(RtpSenderTest, RespectsNackBitrateLimit) {
+  const int32_t kPacketSize = 1400;
+  const int32_t kNumPackets = 30;
+
+  rtp_sender_->SetStorePacketsStatus(true, kNumPackets);
+  rtp_sender_->SetTargetBitrate(kNumPackets * kPacketSize * 8);

[stefan-webrtc, 2015/09/17 11:45:33]

Comment on that you want 30 1400 bytes packets to be possible to send in a
second, just because the unit of what you pass in is in bits and not bits/s.

[sprang_webrtc, 2015/09/21 11:17:05]

Done.

+  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.

[stefan-webrtc, 2015/09/17 11:45:33]

This surprises me a bit. You have already used the bandwidth as far as I can
tell, since you sent 30 packets in the loop above. What am I missing?

[sprang_webrtc, 2015/09/21 11:17:05]

The nack overhead is compensated for in media_optimization when setting new
rates. So this won't kick in until the next OnNetworkChanged. I'll add a
comment.

+  rtp_sender_->OnReceivedNACK(sequence_numbers, 0);
+  EXPECT_EQ(kNumPackets * 2, transport_.packets_sent_);
+
+  // Resending even a single packet should not work, bandwidth exceeded.
+  sequence_numbers.clear();

[stefan-webrtc, 2015/09/17 11:45:33]

This isn't really necessary, as the result should be the same without it.

[sprang_webrtc, 2015/09/21 11:17:05]

Done.

+  sequence_numbers.push_back(kStartSequenceNumber);
+  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;
 
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
                    kRtpExtensionVideoRotation, kVideoRotationExtensionId));
   EXPECT_TRUE(rtp_sender_->ActivateCVORtpHeaderExtension());
 
   EXPECT_EQ(
@@ skipping 12 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