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Add unit test for nack bandwidth constraint.

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BUG=

Review URL: https://codereview.webrtc.org/1341743002

Cr-Commit-Position: refs/heads/master@{#10111}
This commit is contained in:
sprang
2015-09-29 09:48:22 -07:00
committed by Commit bot
parent 98db68fdaa
commit 38778b046f
1 changed files with 29 additions and 0 deletions
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29
webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
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@ -1348,6 +1348,35 @@ 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);
// 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_);
// 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};