Change NetEq::GetAudio to use AudioFrame

webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h

 /*
  *  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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_QUALITY_TEST_H_
 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_QUALITY_TEST_H_
 
 #include <fstream>
 #include <memory>
 #include <gflags/gflags.h>
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
 #include "webrtc/modules/audio_coding/neteq/tools/audio_sink.h"
 #include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
 #include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
 using google::RegisterFlagValidator;
 
 namespace webrtc {
 namespace test {
 
 class LossModel {
  public:
   virtual ~LossModel() {};
@@ skipping 74 matching lines @@
   int decodable_time_ms_;
   double drift_factor_;
   int packet_loss_rate_;
   const int block_duration_ms_;
   const int in_sampling_khz_;
   const int out_sampling_khz_;
 
   // Number of samples per channel in a frame.
   const size_t in_size_samples_;
 
-  // Expected output number of samples per channel in a frame.
-  const size_t out_size_samples_;
-
   size_t payload_size_bytes_;
   size_t max_payload_bytes_;
 
   std::unique_ptr<InputAudioFile> in_file_;
   std::unique_ptr<AudioSink> output_;
   std::ofstream log_file_;
 
   std::unique_ptr<RtpGenerator> rtp_generator_;
   std::unique_ptr<NetEq> neteq_;
   std::unique_ptr<LossModel> loss_model_;
 
   std::unique_ptr<int16_t[]> in_data_;
   rtc::Buffer payload_;
-  std::unique_ptr<int16_t[]> out_data_;
+  AudioFrame out_frame_;
   WebRtcRTPHeader rtp_header_;
 
   size_t total_payload_size_bytes_;
 };
 
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_QUALITY_TEST_H_