Change NetEq::GetAudio to use AudioFrame

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

Index: webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h b/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h
index 43c95b80625e5d7949c04998e116884f338ab316..11afe6886bb781ea953b4ae9c4815e911c7e9420 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h
@@ -19,6 +19,7 @@
 #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;
@@ -113,9 +114,6 @@ class NetEqQualityTest : public ::testing::Test {
   // 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_;
 
@@ -129,7 +127,7 @@ class NetEqQualityTest : public ::testing::Test {
 
   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_;