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

webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.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 114 matching lines @@
 size_t AudioEncoderOpus::Max10MsFramesInAPacket() const {
   return Num10msFramesPerPacket();
 }
 
 int AudioEncoderOpus::GetTargetBitrate() const {
   return config_.bitrate_bps;
 }
 
 AudioEncoder::EncodedInfo AudioEncoderOpus::EncodeInternal(
     uint32_t rtp_timestamp,
-    const int16_t* audio,
+    rtc::ArrayView<const int16_t> audio,
     size_t max_encoded_bytes,
     uint8_t* encoded) {
   if (input_buffer_.empty())
     first_timestamp_in_buffer_ = rtp_timestamp;
-  input_buffer_.insert(input_buffer_.end(), audio,
-                       audio + SamplesPer10msFrame());
+  RTC_DCHECK_EQ(static_cast<size_t>(SamplesPer10msFrame()), audio.size());
+  input_buffer_.insert(input_buffer_.end(), audio.cbegin(), audio.cend());
   if (input_buffer_.size() <
       (static_cast<size_t>(Num10msFramesPerPacket()) * SamplesPer10msFrame())) {
     return EncodedInfo();
   }
   RTC_CHECK_EQ(
       input_buffer_.size(),
       static_cast<size_t>(Num10msFramesPerPacket()) * SamplesPer10msFrame());
   int status = WebRtcOpus_Encode(
       inst_, &input_buffer_[0],
       rtc::CheckedDivExact(input_buffer_.size(),
@@ skipping 97 matching lines @@
     RTC_CHECK_EQ(0, WebRtcOpus_DisableDtx(inst_));
   }
   RTC_CHECK_EQ(0,
                WebRtcOpus_SetPacketLossRate(
                    inst_, static_cast<int32_t>(packet_loss_rate_ * 100 + .5)));
   config_ = config;
   return true;
 }
 
 }  // namespace webrtc