diff --git a/src/modules/rtp_rtcp/source/rtcp_receiver_test.cc b/src/modules/rtp_rtcp/source/rtcp_receiver_test.cc
new file mode 100644
index 000000000..7f55c0e32
--- /dev/null
+++ b/src/modules/rtp_rtcp/source/rtcp_receiver_test.cc
@@ -0,0 +1,199 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file includes unit tests for the RTCPReceiver.
+ */
+#include <gtest/gtest.h>
+
+// Note: This file has no directory. Lint warning must be ignored.
+#include "common_types.h"
+#include "modules/rtp_rtcp/source/rtp_utility.h"
+#include "modules/rtp_rtcp/source/rtcp_sender.h"
+#include "modules/rtp_rtcp/source/rtcp_receiver.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_impl.h"
+
+namespace webrtc {
+
+namespace {  // Anonymous namespace; hide utility functions and classes.
+
+// This test transport verifies that no functions get called.
+class TestTransport : public Transport,
+                      public RtpData {
+ public:
+  explicit TestTransport(RTCPReceiver* rtcp_receiver) :
+    rtcp_receiver_(rtcp_receiver) {
+  }
+
+  virtual int SendPacket(int /*ch*/, const void* /*data*/, int /*len*/) {
+    ADD_FAILURE();  // FAIL() gives a compile error.
+    return -1;
+  }
+
+  // Injects an RTCP packet into the receiver.
+  virtual int SendRTCPPacket(int /* ch */, const void *packet, int packet_len) {
+    ADD_FAILURE();
+    return 0;
+  }
+
+  virtual int OnReceivedPayloadData(const WebRtc_UWord8* payloadData,
+                                    const WebRtc_UWord16 payloadSize,
+                                    const WebRtcRTPHeader* rtpHeader) {
+    ADD_FAILURE();
+    return 0;
+  }
+  RTCPReceiver* rtcp_receiver_;
+};
+
+class RtcpReceiverTest : public ::testing::Test {
+ protected:
+  RtcpReceiverTest() {
+    system_clock_ = ModuleRTPUtility::GetSystemClock();
+    rtp_rtcp_impl_ = new ModuleRtpRtcpImpl(0, false, system_clock_);
+    rtcp_receiver_ = new RTCPReceiver(0, system_clock_, rtp_rtcp_impl_);
+    test_transport_ = new TestTransport(rtcp_receiver_);
+    EXPECT_EQ(0, rtp_rtcp_impl_->RegisterIncomingDataCallback(test_transport_));
+  }
+  ~RtcpReceiverTest() {
+    delete rtcp_sender_;
+    delete rtcp_receiver_;
+    delete rtp_rtcp_impl_;
+    delete test_transport_;
+    delete system_clock_;
+  }
+
+  // Injects an RTCP packet into the receiver.
+  // Returns 0 for OK, non-0 for failure.
+  int InjectRtcpPacket(const WebRtc_UWord8* packet,
+                        WebRtc_UWord16 packet_len) {
+    RTCPUtility::RTCPParserV2 rtcpParser(packet,
+                                         packet_len,
+                                         true);  // Allow non-compound RTCP
+
+    RTCPHelp::RTCPPacketInformation rtcpPacketInformation;
+    int result = rtcp_receiver_->IncomingRTCPPacket(rtcpPacketInformation,
+                                                    &rtcpParser);
+    rtcp_packet_info_ = rtcpPacketInformation;
+    return result;
+  }
+
+  RtpRtcpClock* system_clock_;
+  ModuleRtpRtcpImpl* rtp_rtcp_impl_;
+  RTCPSender* rtcp_sender_;
+  RTCPReceiver* rtcp_receiver_;
+  TestTransport* test_transport_;
+  RTCPHelp::RTCPPacketInformation rtcp_packet_info_;
+};
+
+
+TEST_F(RtcpReceiverTest, BrokenPacketIsIgnored) {
+  const WebRtc_UWord8 bad_packet[] = {0, 0, 0, 0};
+  EXPECT_EQ(0, InjectRtcpPacket(bad_packet, sizeof(bad_packet)));
+  EXPECT_EQ(0U, rtcp_packet_info_.rtcpPacketTypeFlags);
+}
+
+TEST_F(RtcpReceiverTest, InjectSrPacket) {
+  const WebRtc_UWord8 sr_packet[] = {
+    0x81, 200,  // Type 200, report count = 0
+    0, 6,  // length
+    0, 1, 2, 3,  // SSRC of sender
+    0, 1, 2, 3, 4, 5, 6, 7,  // NTP timestamp
+    0, 1, 2, 3,  // RTP timestamp
+    0, 0, 0, 0,  // Sender's packet count
+    0, 0, 0, 0  // Sender's octet count
+  };
+  EXPECT_EQ(0, InjectRtcpPacket(sr_packet, sizeof(sr_packet)));
+  // The parser will note the remote SSRC on a SR from other than his
+  // expected peer, but will not flag that he's gotten a packet.
+  EXPECT_EQ(0x010203U, rtcp_packet_info_.remoteSSRC);
+  EXPECT_EQ(0U,
+            kRtcpSr & rtcp_packet_info_.rtcpPacketTypeFlags);
+}
+
+TEST_F(RtcpReceiverTest, TmmbrReceivedWithNoIncomingPacket) {
+  // This call is expected to fail because no data has arrived.
+  EXPECT_EQ(-1, rtcp_receiver_->TMMBRReceived(0, 0, NULL));
+}
+
+TEST_F(RtcpReceiverTest, TmmbrPacketAccepted) {
+  const WebRtc_UWord8 tmmbr_packet[] = {
+    0x81, 200,  // Type 200 SR, report count = 0
+    0, 6,  // length
+    0, 1, 2, 3,  // SSRC of sender
+    0, 1, 2, 3, 4, 5, 6, 7,  // NTP timestamp
+    0, 1, 2, 3,  // RTP timestamp
+    0, 0, 0, 0,  // Sender's packet count
+    0, 0, 0, 0,  // Sender's octet count
+    // TMMBR
+    0x83, 205,  // Type 205 RTPFB, FMT 3 TMMBR
+    0, 4,  // length
+    0, 1, 2, 3,  // SSRC of sender
+    0, 0, 1, 1,  // SSRC of media source
+    2, 4, 6, 8,  // SSRC we ask to rate-control. Must match "our" SSRC.
+    0, 55, 0, 0   // MxTBR
+  };
+  rtcp_receiver_->SetSSRC(0x2040608);  // Matches "media source" above.
+  EXPECT_EQ(0, InjectRtcpPacket(tmmbr_packet, sizeof(tmmbr_packet)));
+  EXPECT_EQ(1, rtcp_receiver_->TMMBRReceived(0, 0, NULL));
+  TMMBRSet candidate_set;
+  candidate_set.VerifyAndAllocateSet(1);
+  EXPECT_EQ(1, rtcp_receiver_->TMMBRReceived(1, 0, &candidate_set));
+  EXPECT_LT(0U, candidate_set.ptrTmmbrSet[0]);
+  EXPECT_EQ(0x101U, candidate_set.ptrSsrcSet[0]);
+}
+
+TEST_F(RtcpReceiverTest, TmmbrPacketNotForUsIgnored) {
+  const WebRtc_UWord8 tmmbr_packet[] = {
+    0x81, 200,  // Type 200 SR, report count = 0
+    0, 6,   // length
+    0, 1, 2, 3,  // SSRC of sender
+    0, 1, 2, 3, 4, 5, 6, 7,  // NTP timestamp
+    0, 1, 2, 3,  // RTP timestamp
+    0, 0, 0, 0,  // Sender's packet count
+    0, 0, 0, 0,  // Sender's octet count
+    // TMMBR
+    0x83, 205,  // Type 205 RTPFB, FMT 3 TMMBR
+    0, 4,  // length
+    0, 1, 2, 3,  // SSRC of sender
+    0, 0, 1, 1,  // SSRC of media source
+    99, 99, 99, 99,  // SSRC we ask to rate-control. Different from 0x2040608.
+    0, 55, 0, 0   // MxTBR
+  };
+  rtcp_receiver_->SetSSRC(0x2040608);  // Matches "media source" above.
+  EXPECT_EQ(0, InjectRtcpPacket(tmmbr_packet, sizeof(tmmbr_packet)));
+  EXPECT_EQ(0, rtcp_receiver_->TMMBRReceived(0, 0, NULL));
+}
+
+TEST_F(RtcpReceiverTest, TmmbrPacketZeroRateIgnored) {
+  const WebRtc_UWord8 tmmbr_packet[] = {
+    0x81, 200,  // Type 200 SR, report count = 0
+    0, 6,  // length
+    0, 1, 2, 3,  // SSRC of sender
+    0, 1, 2, 3, 4, 5, 6, 7,  // NTP timestamp
+    0, 1, 2, 3,  // RTP timestamp
+    0, 0, 0, 0,  // Sender's packet count
+    0, 0, 0, 0,  // Sender's octet count
+    // TMMBR
+    0x83, 205,  // Type 205 RTPFB, FMT 3 TMMBR
+    0, 4,  // length
+    0, 1, 2, 3,  // SSRC of sender
+    0, 0, 1, 1,  // SSRC of media source
+    2, 4, 6, 8,  // SSRC we ask to rate-control. Must match "our" SSRC.
+    0, 0, 0, 0   // MxTBR == zero
+  };
+  rtcp_receiver_->SetSSRC(0x2040608);  // Matches "media source" above.
+  EXPECT_EQ(0, InjectRtcpPacket(tmmbr_packet, sizeof(tmmbr_packet)));
+  EXPECT_EQ(0, rtcp_receiver_->TMMBRReceived(0, 0, NULL));
+}
+
+}  // Anonymous namespace
+
+}  // namespace webrtc
diff --git a/src/modules/rtp_rtcp/source/rtp_rtcp_tests.gypi b/src/modules/rtp_rtcp/source/rtp_rtcp_tests.gypi
index 1c66efc70..f079a1595 100644
--- a/src/modules/rtp_rtcp/source/rtp_rtcp_tests.gypi
+++ b/src/modules/rtp_rtcp/source/rtp_rtcp_tests.gypi
@@ -36,6 +36,7 @@
         'rtp_header_extension_test.cc',
         'rtp_sender_test.cc',
         'rtcp_sender_test.cc',
+        'rtcp_receiver_test.cc',
         'transmission_bucket_test.cc',
         'vp8_partition_aggregator_unittest.cc',
       ],