diff --git a/webrtc/common_audio/resampler/push_resampler_unittest.cc b/webrtc/common_audio/resampler/push_resampler_unittest.cc
index 4449f4c633..58880cc1b7 100644
--- a/webrtc/common_audio/resampler/push_resampler_unittest.cc
+++ b/webrtc/common_audio/resampler/push_resampler_unittest.cc
@@ -9,6 +9,7 @@
  */
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/checks.h"  // force defintion of RTC_DCHECK_IS_ON
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
 
 // Quality testing of PushResampler is handled through output_mixer_unittest.cc.
@@ -17,12 +18,32 @@ namespace webrtc {
 
 TEST(PushResamplerTest, VerifiesInputParameters) {
   PushResampler<int16_t> resampler;
-  EXPECT_EQ(-1, resampler.InitializeIfNeeded(-1, 16000, 1));
-  EXPECT_EQ(-1, resampler.InitializeIfNeeded(16000, -1, 1));
-  EXPECT_EQ(-1, resampler.InitializeIfNeeded(16000, 16000, 0));
-  EXPECT_EQ(-1, resampler.InitializeIfNeeded(16000, 16000, 3));
   EXPECT_EQ(0, resampler.InitializeIfNeeded(16000, 16000, 1));
   EXPECT_EQ(0, resampler.InitializeIfNeeded(16000, 16000, 2));
 }
 
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+TEST(PushResamplerTest, VerifiesBadInputParameters1) {
+  PushResampler<int16_t> resampler;
+  EXPECT_DEATH(resampler.InitializeIfNeeded(-1, 16000, 1),
+               "src_sample_rate_hz");
+}
+
+TEST(PushResamplerTest, VerifiesBadInputParameters2) {
+  PushResampler<int16_t> resampler;
+  EXPECT_DEATH(resampler.InitializeIfNeeded(16000, -1, 1),
+               "dst_sample_rate_hz");
+}
+
+TEST(PushResamplerTest, VerifiesBadInputParameters3) {
+  PushResampler<int16_t> resampler;
+  EXPECT_DEATH(resampler.InitializeIfNeeded(16000, 16000, 0), "num_channels");
+}
+
+TEST(PushResamplerTest, VerifiesBadInputParameters4) {
+  PushResampler<int16_t> resampler;
+  EXPECT_DEATH(resampler.InitializeIfNeeded(16000, 16000, 3), "num_channels");
+}
+#endif
+
 }  // namespace webrtc