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

webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.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 70 matching lines @@
 size_t AudioEncoderPcm::Max10MsFramesInAPacket() const {
   return num_10ms_frames_per_packet_;
 }
 
 int AudioEncoderPcm::GetTargetBitrate() const {
   return 8 * BytesPerSample() * SampleRateHz() * NumChannels();
 }
 
 AudioEncoder::EncodedInfo AudioEncoderPcm::EncodeInternal(
     uint32_t rtp_timestamp,
-    const int16_t* audio,
+    rtc::ArrayView<const int16_t> audio,
     size_t max_encoded_bytes,
     uint8_t* encoded) {
-  const int num_samples = SampleRateHz() / 100 * NumChannels();
   if (speech_buffer_.empty()) {
     first_timestamp_in_buffer_ = rtp_timestamp;
   }
-  for (int i = 0; i < num_samples; ++i) {
-    speech_buffer_.push_back(audio[i]);
-  }
+  speech_buffer_.insert(speech_buffer_.end(), audio.begin(), audio.end());
   if (speech_buffer_.size() < full_frame_samples_) {
     return EncodedInfo();
   }
   RTC_CHECK_EQ(speech_buffer_.size(), full_frame_samples_);
   RTC_CHECK_GE(max_encoded_bytes, full_frame_samples_);
   EncodedInfo info;
   info.encoded_timestamp = first_timestamp_in_buffer_;
   info.payload_type = payload_type_;
   info.encoded_bytes =
       EncodeCall(&speech_buffer_[0], full_frame_samples_, encoded);
@@ skipping 25 matching lines @@
                                     size_t input_len,
                                     uint8_t* encoded) {
   return WebRtcG711_EncodeU(audio, input_len, encoded);
 }
 
 int AudioEncoderPcmU::BytesPerSample() const {
   return 1;
 }
 
 }  // namespace webrtc