diff --git a/webrtc/video/BUILD.gn b/webrtc/video/BUILD.gn
index 3d34394cdd..a5a3b7c2dd 100644
--- a/webrtc/video/BUILD.gn
+++ b/webrtc/video/BUILD.gn
@@ -204,6 +204,7 @@ if (rtc_include_tests) {
       "send_statistics_proxy_unittest.cc",
       "stats_counter_unittest.cc",
       "stream_synchronization_unittest.cc",
+      "video_receive_stream_unittest.cc",
       "video_send_stream_tests.cc",
       "vie_encoder_unittest.cc",
       "vie_remb_unittest.cc",
diff --git a/webrtc/video/video_receive_stream_unittest.cc b/webrtc/video/video_receive_stream_unittest.cc
new file mode 100644
index 0000000000..6160e284f4
--- /dev/null
+++ b/webrtc/video/video_receive_stream_unittest.cc
@@ -0,0 +1,141 @@
+/*
+ *  Copyright 2017 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.
+ */
+
+#include <vector>
+
+#include "webrtc/test/gtest.h"
+#include "webrtc/test/gmock.h"
+
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/event.h"
+#include "webrtc/media/base/fakevideorenderer.h"
+#include "webrtc/modules/pacing/packet_router.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "webrtc/modules/utility/include/process_thread.h"
+#include "webrtc/video/call_stats.h"
+#include "webrtc/video/video_receive_stream.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/system_wrappers/include/sleep.h"
+#include "webrtc/test/field_trial.h"
+#include "webrtc/video_decoder.h"
+
+using testing::_;
+using testing::Invoke;
+
+constexpr int kDefaultTimeOutMs = 50;
+
+namespace webrtc {
+
+namespace {
+
+const char kNewJitterBufferFieldTrialEnabled[] =
+    "WebRTC-NewVideoJitterBuffer/Enabled/";
+
+class MockTransport : public Transport {
+ public:
+  MOCK_METHOD3(SendRtp,
+               bool(const uint8_t* packet,
+                    size_t length,
+                    const PacketOptions& options));
+  MOCK_METHOD2(SendRtcp, bool(const uint8_t* packet, size_t length));
+};
+
+class MockVideoDecoder : public VideoDecoder {
+ public:
+  MOCK_METHOD2(InitDecode,
+               int32_t(const VideoCodec* config, int32_t number_of_cores));
+  MOCK_METHOD5(Decode,
+               int32_t(const EncodedImage& input,
+                       bool missing_frames,
+                       const RTPFragmentationHeader* fragmentation,
+                       const CodecSpecificInfo* codec_specific_info,
+                       int64_t render_time_ms));
+  MOCK_METHOD1(RegisterDecodeCompleteCallback,
+               int32_t(DecodedImageCallback* callback));
+  MOCK_METHOD0(Release, int32_t(void));
+  const char* ImplementationName() const { return "MockVideoDecoder"; }
+};
+
+}  // namespace
+
+class VideoReceiveStreamTest : public testing::Test {
+ public:
+  VideoReceiveStreamTest()
+      : override_field_trials_(kNewJitterBufferFieldTrialEnabled),
+        config_(&mock_transport_),
+        call_stats_(Clock::GetRealTimeClock()),
+        process_thread_(ProcessThread::Create("TestThread")) {}
+
+  void SetUp() {
+    constexpr int kDefaultNumCpuCores = 2;
+    config_.rtp.remote_ssrc = 1111;
+    config_.rtp.local_ssrc = 2222;
+    config_.renderer = &fake_renderer_;
+    VideoReceiveStream::Decoder h264_decoder;
+    h264_decoder.payload_type = 99;
+    h264_decoder.payload_name = "H264";
+    h264_decoder.codec_params.insert(
+        {"sprop-parameter-sets", "Z0IACpZTBYmI,aMljiA=="});
+    h264_decoder.decoder = &mock_h264_video_decoder_;
+    config_.decoders.push_back(h264_decoder);
+    VideoReceiveStream::Decoder null_decoder;
+    null_decoder.payload_type = 98;
+    null_decoder.payload_name = "null";
+    null_decoder.decoder = &mock_null_video_decoder_;
+    config_.decoders.push_back(null_decoder);
+
+    video_receive_stream_.reset(new webrtc::internal::VideoReceiveStream(
+        kDefaultNumCpuCores,
+        false,  // flex_fec
+        &packet_router_, config_.Copy(), process_thread_.get(), &call_stats_,
+        nullptr));  // remb
+  }
+
+ protected:
+  webrtc::test::ScopedFieldTrials override_field_trials_;
+  VideoReceiveStream::Config config_;
+  CallStats call_stats_;
+  MockVideoDecoder mock_h264_video_decoder_;
+  MockVideoDecoder mock_null_video_decoder_;
+  cricket::FakeVideoRenderer fake_renderer_;
+  MockTransport mock_transport_;
+  PacketRouter packet_router_;
+  std::unique_ptr<ProcessThread> process_thread_;
+  std::unique_ptr<webrtc::internal::VideoReceiveStream> video_receive_stream_;
+};
+
+TEST_F(VideoReceiveStreamTest, CreateFrameFromH264FmtpSpropAndIdr) {
+  constexpr uint8_t idr_nalu[] = {0x05, 0xFF, 0xFF, 0xFF};
+  RtpPacketToSend rtppacket(nullptr);
+  uint8_t* payload = rtppacket.AllocatePayload(sizeof(idr_nalu));
+  memcpy(payload, idr_nalu, sizeof(idr_nalu));
+  rtppacket.SetMarker(true);
+  rtppacket.SetSsrc(1111);
+  rtppacket.SetPayloadType(99);
+  rtppacket.SetSequenceNumber(1);
+  rtppacket.SetTimestamp(0);
+  rtc::Event init_decode_event_(false, false);
+  EXPECT_CALL(mock_h264_video_decoder_, InitDecode(_, _))
+      .WillOnce(Invoke([&init_decode_event_](const VideoCodec* config,
+                                             int32_t number_of_cores) {
+        init_decode_event_.Set();
+        return 0;
+      }));
+  EXPECT_CALL(mock_h264_video_decoder_, RegisterDecodeCompleteCallback(_));
+  video_receive_stream_->Start();
+  EXPECT_CALL(mock_h264_video_decoder_, Decode(_, false, _, _, _));
+  EXPECT_EQ(true,
+            video_receive_stream_->OnRecoveredPacket(rtppacket.data(),
+                                                     rtppacket.size()));
+  EXPECT_CALL(mock_h264_video_decoder_, Release());
+  // Make sure the decoder thread had a chance to run.
+  init_decode_event_.Wait(kDefaultTimeOutMs);
+}
+}  // namespace webrtc