diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
index d30a63c448..eefe0a5420 100644
--- a/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
@@ -309,11 +309,14 @@ TEST_F(AudioCodingModuleTestOldApi, VerifyOutputFrame) {
   EXPECT_EQ(kSampleRateHz, audio_frame.sample_rate_hz_);
 }
 
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
 TEST_F(AudioCodingModuleTestOldApi, FailOnZeroDesiredFrequency) {
   AudioFrame audio_frame;
   bool muted;
-  EXPECT_EQ(-1, acm_->PlayoutData10Ms(0, &audio_frame, &muted));
+  EXPECT_DEATH(acm_->PlayoutData10Ms(0, &audio_frame, &muted),
+               "dst_sample_rate_hz");
 }
+#endif
 
 // Checks that the transport callback is invoked once for each speech packet.
 // Also checks that the frame type is kAudioFrameSpeech.