diff --git a/media/engine/webrtc_video_engine_unittest.cc b/media/engine/webrtc_video_engine_unittest.cc
index 4270e274f5..6d29e3d89a 100644
--- a/media/engine/webrtc_video_engine_unittest.cc
+++ b/media/engine/webrtc_video_engine_unittest.cc
@@ -1417,6 +1417,10 @@ class WebRtcVideoChannelEncodedFrameCallbackTest : public ::testing::Test {
     channel_->SetRecvParameters(parameters);
   }
 
+  ~WebRtcVideoChannelEncodedFrameCallbackTest() override {
+    channel_->SetInterface(nullptr);
+  }
+
   void DeliverKeyFrame(uint32_t ssrc) {
     webrtc::RtpPacket packet;
     packet.SetMarker(true);
@@ -1544,7 +1548,7 @@ class WebRtcVideoChannelBaseTest : public ::testing::Test {
                 webrtc::CreateBuiltinVideoDecoderFactory(),
                 field_trials_) {}
 
-  virtual void SetUp() {
+  void SetUp() override {
     // One testcase calls SetUp in a loop, only create call_ once.
     if (!call_) {
       webrtc::Call::Config call_config(&event_log_);
@@ -1586,6 +1590,7 @@ class WebRtcVideoChannelBaseTest : public ::testing::Test {
     // Make the second renderer available for use by a new stream.
     EXPECT_TRUE(channel_->SetSink(kSsrc + 2, &renderer2_));
   }
+
   // Setup an additional stream just to send video. Defer add recv stream.
   // This is required if you want to test unsignalled recv of video rtp packets.
   void SetUpSecondStreamWithNoRecv() {
@@ -1604,7 +1609,17 @@ class WebRtcVideoChannelBaseTest : public ::testing::Test {
     EXPECT_TRUE(
         channel_->SetVideoSend(kSsrc + 2, nullptr, frame_forwarder_2_.get()));
   }
-  virtual void TearDown() { channel_.reset(); }
+
+  void TearDown() override {
+    channel_->SetInterface(nullptr);
+    channel_.reset();
+  }
+
+  void ResetTest() {
+    TearDown();
+    SetUp();
+  }
+
   bool SetDefaultCodec() { return SetOneCodec(DefaultCodec()); }
 
   bool SetOneCodec(const cricket::VideoCodec& codec) {
@@ -1788,7 +1803,7 @@ TEST_F(WebRtcVideoChannelBaseTest, OverridesRecvBufferSize) {
   const int kCustomRecvBufferSize = 123456;
   webrtc::test::ScopedFieldTrials field_trial(
       "WebRTC-IncreasedReceivebuffers/123456/");
-  SetUp();
+  ResetTest();
 
   EXPECT_TRUE(SetOneCodec(DefaultCodec()));
   EXPECT_TRUE(SetSend(true));
@@ -1804,7 +1819,7 @@ TEST_F(WebRtcVideoChannelBaseTest, OverridesRecvBufferSizeWithSuffix) {
   const int kCustomRecvBufferSize = 123456;
   webrtc::test::ScopedFieldTrials field_trial(
       "WebRTC-IncreasedReceivebuffers/123456_Dogfood/");
-  SetUp();
+  ResetTest();
 
   EXPECT_TRUE(SetOneCodec(DefaultCodec()));
   EXPECT_TRUE(SetSend(true));
@@ -1824,7 +1839,7 @@ TEST_F(WebRtcVideoChannelBaseTest, InvalidRecvBufferSize) {
     field_trial_string += group;
     field_trial_string += "/";
     webrtc::test::ScopedFieldTrials field_trial(field_trial_string);
-    SetUp();
+    ResetTest();
 
     EXPECT_TRUE(SetOneCodec(DefaultCodec()));
     EXPECT_TRUE(SetSend(true));
@@ -1835,8 +1850,6 @@ TEST_F(WebRtcVideoChannelBaseTest, InvalidRecvBufferSize) {
 
 // Test that stats work properly for a 1-1 call.
 TEST_F(WebRtcVideoChannelBaseTest, GetStats) {
-  SetUp();
-
   const int kDurationSec = 3;
   const int kFps = 10;
   SendReceiveManyAndGetStats(DefaultCodec(), kDurationSec, kFps);
@@ -1893,8 +1906,6 @@ TEST_F(WebRtcVideoChannelBaseTest, GetStats) {
 
 // Test that stats work properly for a conf call with multiple recv streams.
 TEST_F(WebRtcVideoChannelBaseTest, GetStatsMultipleRecvStreams) {
-  SetUp();
-
   cricket::FakeVideoRenderer renderer1, renderer2;
   EXPECT_TRUE(SetOneCodec(DefaultCodec()));
   cricket::VideoSendParameters parameters;
@@ -2513,6 +2524,16 @@ class WebRtcVideoChannelTest : public WebRtcVideoEngineTest {
     ASSERT_TRUE(channel_->SetSendParameters(send_parameters_));
   }
 
+  void TearDown() override {
+    channel_->SetInterface(nullptr);
+    channel_ = nullptr;
+  }
+
+  void ResetTest() {
+    TearDown();
+    SetUp();
+  }
+
   cricket::VideoCodec GetEngineCodec(const std::string& name) {
     for (const cricket::VideoCodec& engine_codec : engine_.send_codecs()) {
       if (absl::EqualsIgnoreCase(name, engine_codec.name))
@@ -3165,7 +3186,7 @@ TEST_F(WebRtcVideoChannelTest, LossNotificationIsEnabledByFieldTrial) {
   RTC_DCHECK(!override_field_trials_);
   override_field_trials_ = std::make_unique<webrtc::test::ScopedFieldTrials>(
       "WebRTC-RtcpLossNotification/Enabled/");
-  SetUp();
+  ResetTest();
   TestLossNotificationState(true);
 }
 
@@ -3173,7 +3194,7 @@ TEST_F(WebRtcVideoChannelTest, LossNotificationCanBeEnabledAndDisabled) {
   RTC_DCHECK(!override_field_trials_);
   override_field_trials_ = std::make_unique<webrtc::test::ScopedFieldTrials>(
       "WebRTC-RtcpLossNotification/Enabled/");
-  SetUp();
+  ResetTest();
 
   AssignDefaultCodec();
   VerifyCodecHasDefaultFeedbackParams(default_codec_, true);
@@ -5248,6 +5269,7 @@ TEST_F(WebRtcVideoChannelTest, TestSetDscpOptions) {
   channel->SetInterface(network_interface.get());
   // Default value when DSCP is disabled should be DSCP_DEFAULT.
   EXPECT_EQ(rtc::DSCP_DEFAULT, network_interface->dscp());
+  channel->SetInterface(nullptr);
 
   // Default value when DSCP is enabled is also DSCP_DEFAULT, until it is set
   // through rtp parameters.
@@ -5277,6 +5299,7 @@ TEST_F(WebRtcVideoChannelTest, TestSetDscpOptions) {
   EXPECT_TRUE(static_cast<webrtc::Transport*>(channel.get())
                   ->SendRtcp(kData, sizeof(kData)));
   EXPECT_EQ(rtc::DSCP_CS1, network_interface->options().dscp);
+  channel->SetInterface(nullptr);
 
   // Verify that setting the option to false resets the
   // DiffServCodePoint.
@@ -5287,6 +5310,7 @@ TEST_F(WebRtcVideoChannelTest, TestSetDscpOptions) {
           video_bitrate_allocator_factory_.get())));
   channel->SetInterface(network_interface.get());
   EXPECT_EQ(rtc::DSCP_DEFAULT, network_interface->dscp());
+  channel->SetInterface(nullptr);
 }
 
 // This test verifies that the RTCP reduced size mode is properly applied to
diff --git a/media/engine/webrtc_voice_engine_unittest.cc b/media/engine/webrtc_voice_engine_unittest.cc
index 3286837d81..ecf8c07e6a 100644
--- a/media/engine/webrtc_voice_engine_unittest.cc
+++ b/media/engine/webrtc_voice_engine_unittest.cc
@@ -3203,6 +3203,7 @@ TEST_P(WebRtcVoiceEngineTestFake, TestSetDscpOptions) {
   channel->SetInterface(&network_interface);
   // Default value when DSCP is disabled should be DSCP_DEFAULT.
   EXPECT_EQ(rtc::DSCP_DEFAULT, network_interface.dscp());
+  channel->SetInterface(nullptr);
 
   config.enable_dscp = true;
   channel.reset(static_cast<cricket::WebRtcVoiceMediaChannel*>(
@@ -3229,6 +3230,7 @@ TEST_P(WebRtcVoiceEngineTestFake, TestSetDscpOptions) {
   const uint8_t kData[10] = {0};
   EXPECT_TRUE(channel->SendRtcp(kData, sizeof(kData)));
   EXPECT_EQ(rtc::DSCP_CS1, network_interface.options().dscp);
+  channel->SetInterface(nullptr);
 
   // Verify that setting the option to false resets the
   // DiffServCodePoint.