Chromium Code Reviews| Index: webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc |
| diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc |
| index e09c29d72c39a6c59114132435ed4543ed66ac00..a7a14a6f86b5bb76611b01dabdd9f0e1356cfbb6 100644 |
| --- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc |
| +++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc |
| @@ -19,11 +19,13 @@ |
| #include "webrtc/common_types.h" |
| #include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_observer.h" |
| #include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h" |
| +#include "webrtc/modules/rtp_rtcp/source/byte_io.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" |
| +#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h" |
| namespace webrtc { |
| @@ -120,9 +122,10 @@ class RtcpReceiverTest : public ::testing::Test { |
| rtcp_packet_info_.ntp_frac = rtcpPacketInformation.ntp_frac; |
| rtcp_packet_info_.rtp_timestamp = rtcpPacketInformation.rtp_timestamp; |
| rtcp_packet_info_.xr_dlrr_item = rtcpPacketInformation.xr_dlrr_item; |
| - if (rtcpPacketInformation.VoIPMetric) { |
| + if (rtcpPacketInformation.VoIPMetric) |
| rtcp_packet_info_.AddVoIPMetric(rtcpPacketInformation.VoIPMetric); |
| - } |
| + rtcp_packet_info_.transport_feedback_.reset( |
| + rtcpPacketInformation.transport_feedback_.release()); |
| return 0; |
| } |
| @@ -1028,6 +1031,69 @@ TEST_F(RtcpReceiverTest, Callbacks) { |
| kCumulativeLoss, kJitter)); |
| } |
| +TEST_F(RtcpReceiverTest, ReceivesTransportFeedback) { |
| + const uint32_t kSenderSsrc = 0x10203; |
| + const uint32_t kSourceSsrc = 0x123456; |
| + |
| + std::set<uint32_t> ssrcs; |
| + ssrcs.insert(kSourceSsrc); |
| + rtcp_receiver_->SetSsrcs(kSourceSsrc, ssrcs); |
| + |
| + // Send a compound packet with a TransportFeedback followed by something else. |
|
stefan-webrtc
2015/09/30 10:45:27
Where's the "followed by something else" part? I o
sprang_webrtc
2015/09/30 11:09:27
Copy/paste mistake. Removed comment.
|
| + rtcp::TransportFeedback packet; |
| + packet.WithMediaSourceSsrc(kSourceSsrc); |
| + packet.WithPacketSenderSsrc(kSenderSsrc); |
| + packet.WithBase(1, 1000); |
| + packet.WithReceivedPacket(1, 1000); |
| + |
| + rtc::scoped_ptr<rtcp::RawPacket> built_packet = packet.Build(); |
| + ASSERT_TRUE(built_packet.get() != nullptr); |
| + |
| + EXPECT_EQ(0, |
| + InjectRtcpPacket(built_packet->Buffer(), built_packet->Length())); |
| + |
| + EXPECT_NE(0u, rtcp_packet_info_.rtcpPacketTypeFlags & kRtcpTransportFeedback); |
| + EXPECT_TRUE(rtcp_packet_info_.transport_feedback_.get() != nullptr); |
| +} |
| + |
| +TEST_F(RtcpReceiverTest, HandlesInvalidTransportFeedback) { |
| + const uint32_t kSenderSsrc = 0x10203; |
| + const uint32_t kSourceSsrc = 0x123456; |
| + |
| + std::set<uint32_t> ssrcs; |
| + ssrcs.insert(kSourceSsrc); |
| + rtcp_receiver_->SetSsrcs(kSourceSsrc, ssrcs); |
| + |
| + // Send a compound packet with a TransportFeedback followed by something else. |
| + rtcp::TransportFeedback packet; |
| + packet.WithMediaSourceSsrc(kSourceSsrc); |
| + packet.WithPacketSenderSsrc(kSenderSsrc); |
| + packet.WithBase(1, 1000); |
| + packet.WithReceivedPacket(1, 1000); |
| + |
| + static uint32_t kBitrateBps = 50000; |
| + rtcp::Remb remb; |
| + remb.From(kSourceSsrc); |
| + remb.WithBitrateBps(kBitrateBps); |
| + packet.Append(&remb); |
| + |
| + rtc::scoped_ptr<rtcp::RawPacket> built_packet = packet.Build(); |
| + ASSERT_TRUE(built_packet.get() != nullptr); |
| + |
| + // Modify the TransportFeedback packet so that it is invalid. |
| + const size_t kStatusCountOffset = 14; |
| + ByteWriter<uint16_t>::WriteBigEndian( |
| + &built_packet->MutableBuffer()[kStatusCountOffset], 42); |
|
stefan-webrtc
2015/09/30 10:45:27
Sounds like we may want to fuzz this code. Maybe a
sprang_webrtc
2015/09/30 11:09:27
Acknowledged.
|
| + |
| + EXPECT_EQ(0, |
| + InjectRtcpPacket(built_packet->Buffer(), built_packet->Length())); |
| + |
| + // Transport feedback should be ignored, but next packet should work. |
| + EXPECT_EQ(0u, rtcp_packet_info_.rtcpPacketTypeFlags & kRtcpTransportFeedback); |
| + EXPECT_NE(0u, rtcp_packet_info_.rtcpPacketTypeFlags & kRtcpRemb); |
| + EXPECT_EQ(kBitrateBps, rtcp_packet_info_.receiverEstimatedMaxBitrate); |
| +} |
| + |
| } // Anonymous namespace |
| } // namespace webrtc |