Adding a some checks and switching out a few assert for RTC_[D]CHECK.

webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc

 /*
  *  Copyright (c) 2014 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.
  */
 
@@ skipping 291 matching lines @@
   EXPECT_EQ(0, acm_->PlayoutData10Ms(kSampleRateHz, &audio_frame, &muted));
   ASSERT_FALSE(muted);
   EXPECT_EQ(id_, audio_frame.id_);
   EXPECT_EQ(0u, audio_frame.timestamp_);
   EXPECT_GT(audio_frame.num_channels_, 0u);
   EXPECT_EQ(static_cast<size_t>(kSampleRateHz / 100),
             audio_frame.samples_per_channel_);
   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.
 TEST_F(AudioCodingModuleTestOldApi, TransportCallbackIsInvokedForEachPacket) {
   const int k10MsBlocksPerPacket = 3;
   codec_.pacsize = k10MsBlocksPerPacket * kSampleRateHz / 100;
   RegisterCodec();
   const int kLoops = 10;
   for (int i = 0; i < kLoops; ++i) {
     EXPECT_EQ(i / k10MsBlocksPerPacket, packet_cb_.num_calls());
@@ skipping 1462 matching lines @@
   Run(16000, 8000, 1000);
 }
 
 TEST_F(AcmSwitchingOutputFrequencyOldApi, Toggle8KhzTo16Khz) {
   Run(8000, 16000, 1000);
 }
 
 #endif
 
 }  // namespace webrtc