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 409be1a66ce16ad7fb9b6f5c4559eadf5f154ae3..fc834201c67614d22bab660e229926f05e115eb8 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc |
@@ -1348,6 +1348,34 @@ TEST_F(RtpSenderTest, BytesReportedCorrectly) { |
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
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
sprang_webrtc
2015/09/21 11:17:05
The nack overhead is compensated for in media_opti
|
+ 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
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}; |