Change NetEq::GetAudio to use AudioFrame

webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.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.
  */
 
 #include <memory>
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h"
 #include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 namespace test {
 
 using ::testing::_;
 using ::testing::SetArgPointee;
 using ::testing::Return;
 
 class MockAudioDecoder final : public AudioDecoder {
  public:
@@ skipping 50 matching lines @@
 
  private:
   const size_t num_channels_;
   bool fec_enabled_;
 };
 
 class NetEqNetworkStatsTest : public NetEqExternalDecoderTest {
  public:
   static const int kPayloadSizeByte = 30;
   static const int kFrameSizeMs = 20;
-  static const int kMaxOutputSize = 960;  // 10 ms * 48 kHz * 2 channels.
 
 enum logic {
   kIgnore,
   kEqual,
   kSmallerThan,
   kLargerThan,
 };
 
 struct NetEqNetworkStatsCheck {
   logic current_buffer_size_ms;
@@ skipping 91 matching lines @@
     for (int k = 0; k < num_loops; ++k) {
       // Delay by one frame such that the FEC can come in.
       while (time_now + kFrameSizeMs >= next_send_time) {
         next_send_time = rtp_generator_->GetRtpHeader(kPayloadType,
                                                       frame_size_samples_,
                                                       &rtp_header_);
         if (!Lost(next_send_time)) {
           InsertPacket(rtp_header_, payload_, next_send_time);
         }
       }
-      GetOutputAudio(kMaxOutputSize, output_, &output_type);
+      GetOutputAudio(&output_frame_, &output_type);
       time_now += kOutputLengthMs;
     }
     CheckNetworkStatistics(expects);
     neteq()->FlushBuffers();
   }
 
   void DecodeFecTest() {
     external_decoder_->set_fec_enabled(false);
     NetEqNetworkStatsCheck expects = {
       kIgnore,  // current_buffer_size_ms
@@ skipping 53 matching lines @@
 
  private:
   MockAudioDecoder* external_decoder_;
   const int samples_per_ms_;
   const size_t frame_size_samples_;
   std::unique_ptr<test::RtpGenerator> rtp_generator_;
   WebRtcRTPHeader rtp_header_;
   uint32_t last_lost_time_;
   uint32_t packet_loss_interval_;
   uint8_t payload_[kPayloadSizeByte];
-  int16_t output_[kMaxOutputSize];
+  AudioFrame output_frame_;
 };
 
 TEST(NetEqNetworkStatsTest, DecodeFec) {
   MockAudioDecoder decoder(1);
   NetEqNetworkStatsTest test(NetEqDecoder::kDecoderOpus, &decoder);
   test.DecodeFecTest();
   EXPECT_CALL(decoder, Die()).Times(1);
 }
 
 TEST(NetEqNetworkStatsTest, StereoDecodeFec) {
   MockAudioDecoder decoder(2);
   NetEqNetworkStatsTest test(NetEqDecoder::kDecoderOpus, &decoder);
   test.DecodeFecTest();
   EXPECT_CALL(decoder, Die()).Times(1);
 }
 
 TEST(NetEqNetworkStatsTest, NoiseExpansionTest) {
   MockAudioDecoder decoder(1);
   NetEqNetworkStatsTest test(NetEqDecoder::kDecoderOpus, &decoder);
   test.NoiseExpansionTest();
   EXPECT_CALL(decoder, Die()).Times(1);
 }
 
 }  // namespace test
 }  // namespace webrtc