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

webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h

 /*
  *  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 14 matching lines @@
   MOCK_CONST_METHOD0(MaxEncodedBytes, size_t());
   MOCK_CONST_METHOD0(SampleRateHz, int());
   MOCK_CONST_METHOD0(NumChannels, int());
   MOCK_CONST_METHOD0(RtpTimestampRateHz, int());
   MOCK_CONST_METHOD0(Num10MsFramesInNextPacket, size_t());
   MOCK_CONST_METHOD0(Max10MsFramesInAPacket, size_t());
   MOCK_CONST_METHOD0(GetTargetBitrate, int());
   // Note, we explicitly chose not to create a mock for the Encode method.
   MOCK_METHOD4(EncodeInternal,
                EncodedInfo(uint32_t timestamp,
-                           const int16_t* audio,
+                           rtc::ArrayView<const int16_t> audio,
                            size_t max_encoded_bytes,
                            uint8_t* encoded));
   MOCK_METHOD0(Reset, void());
   MOCK_METHOD1(SetFec, bool(bool enable));
   MOCK_METHOD1(SetDtx, bool(bool enable));
   MOCK_METHOD1(SetApplication, bool(Application application));
   MOCK_METHOD1(SetMaxPlaybackRate, void(int frequency_hz));
   MOCK_METHOD1(SetProjectedPacketLossRate, void(double fraction));
   MOCK_METHOD1(SetTargetBitrate, void(int target_bps));
   MOCK_METHOD1(SetMaxBitrate, void(int max_bps));
   MOCK_METHOD1(SetMaxPayloadSize, void(int max_payload_size_bytes));
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_MOCK_MOCK_AUDIO_ENCODER_H_