Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(413)

Unified Diff: webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc

Issue 1367193002: Fix bug where rtcp::TransportFeedback may generate incorrect messages. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Addressed comments Created 5 years, 3 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
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

Powered by Google App Engine
This is Rietveld 408576698