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

webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.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 79 matching lines @@
 size_t AudioEncoderCng::Max10MsFramesInAPacket() const {
   return speech_encoder_->Max10MsFramesInAPacket();
 }
 
 int AudioEncoderCng::GetTargetBitrate() const {
   return speech_encoder_->GetTargetBitrate();
 }
 
 AudioEncoder::EncodedInfo AudioEncoderCng::EncodeInternal(
     uint32_t rtp_timestamp,
-    const int16_t* audio,
+    rtc::ArrayView<const int16_t> audio,
     size_t max_encoded_bytes,
     uint8_t* encoded) {
   RTC_CHECK_GE(max_encoded_bytes,
                static_cast<size_t>(num_cng_coefficients_ + 1));
   const size_t samples_per_10ms_frame = SamplesPer10msFrame();
   RTC_CHECK_EQ(speech_buffer_.size(),
                rtp_timestamps_.size() * samples_per_10ms_frame);
   rtp_timestamps_.push_back(rtp_timestamp);
-  for (size_t i = 0; i < samples_per_10ms_frame; ++i) {
-    speech_buffer_.push_back(audio[i]);
-  }
+  RTC_DCHECK_EQ(samples_per_10ms_frame, audio.size());
+  speech_buffer_.insert(speech_buffer_.end(), audio.cbegin(), audio.cend());
   const size_t frames_to_encode = speech_encoder_->Num10MsFramesInNextPacket();
   if (rtp_timestamps_.size() < frames_to_encode) {
     return EncodedInfo();
   }
   RTC_CHECK_LE(static_cast<int>(frames_to_encode * 10), kMaxFrameSizeMs)
       << "Frame size cannot be larger than " << kMaxFrameSizeMs
       << " ms when using VAD/CNG.";
 
   // Group several 10 ms blocks per VAD call. Call VAD once or twice using the
   // following split sizes:
@@ skipping 113 matching lines @@
   return info;
 }
 
 AudioEncoder::EncodedInfo AudioEncoderCng::EncodeActive(
     size_t frames_to_encode,
     size_t max_encoded_bytes,
     uint8_t* encoded) {
   const size_t samples_per_10ms_frame = SamplesPer10msFrame();
   AudioEncoder::EncodedInfo info;
   for (size_t i = 0; i < frames_to_encode; ++i) {
-    info = speech_encoder_->Encode(
-        rtp_timestamps_.front(), &speech_buffer_[i * samples_per_10ms_frame],
-        samples_per_10ms_frame, max_encoded_bytes, encoded);
+    info =
+        speech_encoder_->Encode(rtp_timestamps_.front(),
+                                rtc::ArrayView<const int16_t>(
+                                    &speech_buffer_[i * samples_per_10ms_frame],
+                                    samples_per_10ms_frame),
+                                max_encoded_bytes, encoded);
     if (i + 1 == frames_to_encode) {
       RTC_CHECK_GT(info.encoded_bytes, 0u) << "Encoder didn't deliver data.";
     } else {
       RTC_CHECK_EQ(info.encoded_bytes, 0u)
           << "Encoder delivered data too early.";
     }
   }
   return info;
 }
 
 size_t AudioEncoderCng::SamplesPer10msFrame() const {
   return rtc::CheckedDivExact(10 * SampleRateHz(), 1000);
 }
 
 }  // namespace webrtc