Change NetEq::GetAudio to use AudioFrame

webrtc/modules/audio_coding/neteq/audio_multi_vector_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 190 matching lines @@
 
   // Read too many samples. Expect to get all samples from the vector.
   EXPECT_EQ(interleaved_length_,
             vec.ReadInterleaved(array_length() + 1, output));
   EXPECT_EQ(0,
             memcmp(array_interleaved_, output, read_samples * sizeof(int16_t)));
 
   delete [] output;
 }
 
-// Try to read to a NULL pointer. Expected to return 0.
-TEST_P(AudioMultiVectorTest, ReadInterleavedToNull) {
-  AudioMultiVector vec(num_channels_);
-  vec.PushBackInterleaved(array_interleaved_, interleaved_length_);
-  int16_t* output = NULL;
-  // Read 5 samples.
-  size_t read_samples = 5;
-  EXPECT_EQ(0u, vec.ReadInterleaved(read_samples, output));
-}
-
 // Test the PopFront method.
 TEST_P(AudioMultiVectorTest, PopFront) {
   AudioMultiVector vec(num_channels_);
   vec.PushBackInterleaved(array_interleaved_, interleaved_length_);
   vec.PopFront(1);  // Remove one element from each channel.
   ASSERT_EQ(array_length() - 1u, vec.Size());
   // Let |ptr| point to the second element of the first channel in the
   // interleaved array.
   int16_t* ptr = &array_interleaved_[num_channels_];
   for (size_t i = 0; i < array_length() - 1; ++i) {
@@ skipping 93 matching lines @@
     EXPECT_EQ(vec[0][i], vec[num_channels_ - 1][i]);
   }
 }
 
 INSTANTIATE_TEST_CASE_P(TestNumChannels,
                         AudioMultiVectorTest,
                         ::testing::Values(static_cast<size_t>(1),
                                           static_cast<size_t>(2),
                                           static_cast<size_t>(5)));
 }  // namespace webrtc