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..0bb3e14f0a7f18823d2e8ddaf366c85cbff09ba3 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 +1031,68 @@ 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); |
+ |
+ 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); |
+ |
+ 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 |