Pass audio to AudioEncoder::Encode() in an ArrayView

webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.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 42 matching lines @@
   void TearDown() override {
     red_.reset();
     // Don't expect the red_ object to delete the AudioEncoder object. But it
     // will be deleted with the test fixture. This is why we explicitly delete
     // the red_ object above, and set expectations on mock_encoder_ afterwards.
     EXPECT_CALL(mock_encoder_, Die()).Times(1);
   }
 
   void Encode() {
     ASSERT_TRUE(red_.get() != NULL);
-    encoded_info_ = red_->Encode(timestamp_, audio_, num_audio_samples_10ms,
-                                 encoded_.size(), &encoded_[0]);
+    encoded_info_ = red_->Encode(
+        timestamp_,
+        rtc::ArrayView<const int16_t>(audio_, num_audio_samples_10ms),
+        encoded_.size(), &encoded_[0]);
     timestamp_ += num_audio_samples_10ms;
   }
 
   MockAudioEncoder mock_encoder_;
   rtc::scoped_ptr<AudioEncoderCopyRed> red_;
   uint32_t timestamp_;
   int16_t audio_[kMaxNumSamples];
   const int sample_rate_hz_;
   size_t num_audio_samples_10ms;
   std::vector<uint8_t> encoded_;
   AudioEncoder::EncodedInfo encoded_info_;
   const int red_payload_type_;
 };
 
 class MockEncodeHelper {
  public:
   MockEncodeHelper() : write_payload_(false), payload_(NULL) {
     memset(&info_, 0, sizeof(info_));
   }
 
   AudioEncoder::EncodedInfo Encode(uint32_t timestamp,
-                                   const int16_t* audio,
+                                   rtc::ArrayView<const int16_t> audio,
                                    size_t max_encoded_bytes,
                                    uint8_t* encoded) {
     if (write_payload_) {
       RTC_CHECK(encoded);
       RTC_CHECK_LE(info_.encoded_bytes, max_encoded_bytes);
       memcpy(encoded, payload_, info_.encoded_bytes);
     }
     return info_;
   }
 
@@ skipping 228 matching lines @@
   config.speech_encoder = NULL;
   EXPECT_DEATH(red = new AudioEncoderCopyRed(config),
                "Speech encoder not provided.");
   // The delete operation is needed to avoid leak reports from memcheck.
   delete red;
 }
 
 #endif  // GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
 
 }  // namespace webrtc