Fix segmentation fault in AudioEncoderOpusTest.EncodeAtMinBitrate.

webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc

 /*
  *  Copyright (c) 2015 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 107 matching lines @@
 // Create 10ms audio data blocks for a total packet size of "packet_size_ms".
 std::unique_ptr<test::AudioLoop> Create10msAudioBlocks(
     const std::unique_ptr<AudioEncoderOpus>& encoder,
     int packet_size_ms) {
   const std::string file_name =
       test::ResourcePath("audio_coding/testfile32kHz", "pcm");
 
   std::unique_ptr<test::AudioLoop> speech_data(new test::AudioLoop());
   int audio_samples_per_ms =
       rtc::CheckedDivExact(encoder->SampleRateHz(), 1000);
-  RTC_DCHECK(speech_data->Init(
-      file_name,
-      packet_size_ms * audio_samples_per_ms * encoder->num_channels_to_encode(),
-      10 * audio_samples_per_ms * encoder->num_channels_to_encode()));
+  if (!speech_data->Init(
+          file_name,
+          packet_size_ms * audio_samples_per_ms *
+              encoder->num_channels_to_encode(),
+          10 * audio_samples_per_ms * encoder->num_channels_to_encode()))
+    return nullptr;
   return speech_data;
 }
 
 }  // namespace
 
 TEST(AudioEncoderOpusTest, DefaultApplicationModeMono) {
   auto states = CreateCodec(1);
   EXPECT_EQ(AudioEncoderOpus::kVoip, states.encoder->application());
 }
 
@@ skipping 372 matching lines @@
     states.encoder->Encode(
         0, rtc::ArrayView<const int16_t>(audio.data(), audio.size()), &encoded);
   }
 }
 
 TEST(AudioEncoderOpusTest, EncodeAtMinBitrate) {
   auto states = CreateCodec(1);
   constexpr int kNumPacketsToEncode = 2;
   auto audio_frames =
       Create10msAudioBlocks(states.encoder, kNumPacketsToEncode * 20);
+  ASSERT_TRUE(audio_frames) << "Create10msAudioBlocks failed";
   rtc::Buffer encoded;
   uint32_t rtp_timestamp = 12345;  // Just a number not important to this test.
 
   states.encoder->OnReceivedUplinkBandwidth(0, rtc::Optional<int64_t>());
   for (int packet_index = 0; packet_index < kNumPacketsToEncode;
        packet_index++) {
     // Make sure we are not encoding before we have enough data for
     // a 20ms packet.
     for (int index = 0; index < 1; index++) {
       states.encoder->Encode(rtp_timestamp, audio_frames->GetNextBlock(),
                              &encoded);
       EXPECT_EQ(0u, encoded.size());
     }
 
     // Should encode now.
     states.encoder->Encode(rtp_timestamp, audio_frames->GetNextBlock(),
                            &encoded);
     EXPECT_GT(encoded.size(), 0u);
     encoded.Clear();
   }
 }
 
 }  // namespace webrtc