Changed AudioEncoder::Encode to take an rtc::Buffer* instead of uint8_t* and a maximum size.

webrtc/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc

 /*
  *  Copyright (c) 2015 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 42 matching lines @@
   void SetUp() override {
     ASSERT_EQ(1u, channels_) << "iLBC supports only mono audio.";
     AudioEncoderIlbc::Config config;
     config.frame_size_ms = FLAGS_frame_size_ms;
     encoder_.reset(new AudioEncoderIlbc(config));
     NetEqQualityTest::SetUp();
   }
 
   int EncodeBlock(int16_t* in_data,
                   size_t block_size_samples,
-                  uint8_t* payload,
-                  size_t max_bytes) override {
+                  rtc::Buffer* payload, size_t max_bytes) override {
     const size_t kFrameSizeSamples = 80;  // Samples per 10 ms.
     size_t encoded_samples = 0;
     uint32_t dummy_timestamp = 0;
     AudioEncoder::EncodedInfo info;
     do {
       info = encoder_->Encode(dummy_timestamp,
                               rtc::ArrayView<const int16_t>(
                                   in_data + encoded_samples, kFrameSizeSamples),
-                              max_bytes, payload);
+                              payload);
       encoded_samples += kFrameSizeSamples;
     } while (info.encoded_bytes == 0);
     return rtc::checked_cast<int>(info.encoded_bytes);
   }
 
  private:
   std::unique_ptr<AudioEncoderIlbc> encoder_;
 };
 
 TEST_F(NetEqIlbcQualityTest, Test) {
   Simulate();
 }
 
 }  // namespace test
 }  // namespace webrtc