Change NetEq::GetAudio to use AudioFrame

webrtc/modules/audio_coding/neteq/audio_multi_vector.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.
  */
 
 #include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
 
 #include <assert.h>
 
 #include <algorithm>
 
+#include "webrtc/base/checks.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 AudioMultiVector::AudioMultiVector(size_t N) {
   assert(N > 0);
   if (N < 1) N = 1;
   for (size_t n = 0; n < N; ++n) {
     channels_.push_back(new AudioVector);
   }
@@ skipping 96 matching lines @@
 }
 
 size_t AudioMultiVector::ReadInterleaved(size_t length,
                                          int16_t* destination) const {
   return ReadInterleavedFromIndex(0, length, destination);
 }
 
 size_t AudioMultiVector::ReadInterleavedFromIndex(size_t start_index,
                                                   size_t length,
                                                   int16_t* destination) const {
-  if (!destination) {
-    return 0;
-  }
+  RTC_DCHECK(destination);
   size_t index = 0;  // Number of elements written to |destination| so far.
   assert(start_index <= Size());
   start_index = std::min(start_index, Size());
   if (length + start_index > Size()) {
     length = Size() - start_index;
   }
   if (num_channels_ == 1) {
     // Special case to avoid the nested for loop below.
     memcpy(destination, &(*this)[0][start_index], length * sizeof(int16_t));
     return length;
@@ skipping 68 matching lines @@
 
 const AudioVector& AudioMultiVector::operator[](size_t index) const {
   return *(channels_[index]);
 }
 
 AudioVector& AudioMultiVector::operator[](size_t index) {
   return *(channels_[index]);
 }
 
 }  // namespace webrtc