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

webrtc/modules/audio_coding/neteq/neteq_unittest.cc

 /*
  *  Copyright (c) 2011 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 921 matching lines @@
     uint8_t payload[kBlockSize32kHz * sizeof(int16_t)];
     WebRtcRTPHeader rtp_info;
     PopulateRtpInfo(0, 0, &rtp_info);
     rtp_info.header.payloadType = payload_type;
 
     int number_channels = 0;
     size_t samples_per_channel = 0;
 
     uint32_t receive_timestamp = 0;
     for (int n = 0; n < 10; ++n) {  // Insert few packets and get audio.
-      size_t enc_len_bytes = WebRtcPcm16b_Encode(
-          input.GetNextBlock(), expected_samples_per_channel, payload);
+      auto block = input.GetNextBlock();
+      ASSERT_EQ(expected_samples_per_channel, block.size());
+      size_t enc_len_bytes =
+          WebRtcPcm16b_Encode(block.data(), block.size(), payload);
       ASSERT_EQ(enc_len_bytes, expected_samples_per_channel * 2);
 
       number_channels = 0;
       samples_per_channel = 0;
       ASSERT_EQ(0,
                 neteq_->InsertPacket(rtp_info, payload, enc_len_bytes,
                                      receive_timestamp));
       ASSERT_EQ(0,
                 neteq_->GetAudio(kBlockSize32kHz,
                                  output,
@@ skipping 577 matching lines @@
     // Pull audio once.
     ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
                                   &num_channels, &type));
     ASSERT_EQ(kBlockSize16kHz, out_len);
   }
   // Verify speech output.
   EXPECT_EQ(kOutputNormal, type);
 }
 
 }  // namespace webrtc