Change NetEq::GetAudio to use AudioFrame

webrtc/modules/audio_coding/neteq/sync_buffer_unittest.cc

 /*
  *  Copyright (c) 2012 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 122 matching lines @@
       new_data[channel][i] = i;
     }
   }
   // Push back |new_data| into |sync_buffer|. This operation should pop out
   // data from the front of |sync_buffer|, so that the size of the buffer
   // remains the same. The |next_index_| should also move with the same length.
   sync_buffer.PushBack(new_data);
 
   // Read to interleaved output. Read in two batches, where each read operation
   // should automatically update the |net_index_| in the SyncBuffer.
-  int16_t output[kChannels * kNewLen];
   // Note that |samples_read| is the number of samples read from each channel.
   // That is, the number of samples written to |output| is
   // |samples_read| * |kChannels|.
-  size_t samples_read = sync_buffer.GetNextAudioInterleaved(kNewLen / 2,
-                                                            output);
-  samples_read +=
-      sync_buffer.GetNextAudioInterleaved(kNewLen / 2,
-                                          &output[samples_read * kChannels]);
-  EXPECT_EQ(kNewLen, samples_read);
+  AudioFrame output1;
+  sync_buffer.GetNextAudioInterleaved(kNewLen / 2, &output1);
+  EXPECT_EQ(kChannels, output1.num_channels_);
+  EXPECT_EQ(kNewLen / 2, output1.samples_per_channel_);
+
+  AudioFrame output2;
+  sync_buffer.GetNextAudioInterleaved(kNewLen / 2, &output2);
+  EXPECT_EQ(kChannels, output2.num_channels_);
+  EXPECT_EQ(kNewLen / 2, output2.samples_per_channel_);
 
   // Verify the data.
-  int16_t* output_ptr = output;
-  for (size_t i = 0; i < kNewLen; ++i) {
+  int16_t* output_ptr = output1.data_;
+  for (size_t i = 0; i < kNewLen / 2; ++i) {
     for (size_t channel = 0; channel < kChannels; ++channel) {
       EXPECT_EQ(new_data[channel][i], *output_ptr);
       ++output_ptr;
+    }
+  }
+  output_ptr = output2.data_;
+  for (size_t i = kNewLen / 2; i < kNewLen; ++i) {
+    for (size_t channel = 0; channel < kChannels; ++channel) {
+      EXPECT_EQ(new_data[channel][i], *output_ptr);
+      ++output_ptr;
     }
   }
 }
 
 }  // namespace webrtc