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

webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.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.
  */
 
 #include "webrtc/modules/audio_coding/codecs/ilbc/include/audio_encoder_ilbc.h"
 
-#include <cstring>
+#include <algorithm>
 #include <limits>
 #include "webrtc/base/checks.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/codecs/ilbc/include/ilbc.h"
 
 namespace webrtc {
 
 namespace {
 
 const int kSampleRateHz = 8000;
@@ skipping 60 matching lines @@
     case 3: case 6:
       // 50 bytes per frame of 30 ms => (approx) 13333 bits/s.
       return 13333;
     default:
       FATAL();
   }
 }
 
 AudioEncoder::EncodedInfo AudioEncoderIlbc::EncodeInternal(
     uint32_t rtp_timestamp,
-    const int16_t* audio,
+    rtc::ArrayView<const int16_t> audio,
     size_t max_encoded_bytes,
     uint8_t* encoded) {
   RTC_DCHECK_GE(max_encoded_bytes, RequiredOutputSizeBytes());
 
   // Save timestamp if starting a new packet.
   if (num_10ms_frames_buffered_ == 0)
     first_timestamp_in_buffer_ = rtp_timestamp;
 
   // Buffer input.
-  std::memcpy(input_buffer_ + kSampleRateHz / 100 * num_10ms_frames_buffered_,
-              audio,
-              kSampleRateHz / 100 * sizeof(audio[0]));
+  RTC_DCHECK_EQ(static_cast<size_t>(kSampleRateHz / 100), audio.size());
+  std::copy(audio.cbegin(), audio.cend(),
+            input_buffer_ + kSampleRateHz / 100 * num_10ms_frames_buffered_);
 
   // If we don't yet have enough buffered input for a whole packet, we're done
   // for now.
   if (++num_10ms_frames_buffered_ < num_10ms_frames_per_packet_) {
     return EncodedInfo();
   }
 
   // Encode buffered input.
   RTC_DCHECK_EQ(num_10ms_frames_buffered_, num_10ms_frames_per_packet_);
   num_10ms_frames_buffered_ = 0;
@@ skipping 27 matching lines @@
   switch (num_10ms_frames_per_packet_) {
     case 2:   return 38;
     case 3:   return 50;
     case 4:   return 2 * 38;
     case 6:   return 2 * 50;
     default:  FATAL();
   }
 }
 
 }  // namespace webrtc