Change NetEq::GetAudio to use AudioFrame

webrtc/modules/audio_coding/neteq/sync_buffer.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 <assert.h>
-
 #include <algorithm>  // Access to min.
 
+#include "webrtc/base/checks.h"
 #include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
 
 namespace webrtc {
 
 size_t SyncBuffer::FutureLength() const {
   return Size() - next_index_;
 }
 
 void SyncBuffer::PushBack(const AudioMultiVector& append_this) {
   size_t samples_added = append_this.Size();
@@ skipping 39 matching lines @@
   position = std::min(position, Size());  // Cap |position| in the valid range.
   length = std::min(length, Size() - position);
   AudioMultiVector::OverwriteAt(insert_this, length, position);
 }
 
 void SyncBuffer::ReplaceAtIndex(const AudioMultiVector& insert_this,
                                 size_t position) {
   ReplaceAtIndex(insert_this, insert_this.Size(), position);
 }
 
-size_t SyncBuffer::GetNextAudioInterleaved(size_t requested_len,
-                                           int16_t* output) {
-  if (!output) {
-    assert(false);
-    return 0;
-  }
-  size_t samples_to_read = std::min(FutureLength(), requested_len);
-  ReadInterleavedFromIndex(next_index_, samples_to_read, output);
-  next_index_ += samples_to_read;
-  return samples_to_read;
+void SyncBuffer::GetNextAudioInterleaved(size_t requested_len,
+                                         AudioFrame* output) {
+  RTC_DCHECK(output);
+  const size_t samples_to_read = std::min(FutureLength(), requested_len);
+  output->Reset();
+  const size_t tot_samples_read =
+      ReadInterleavedFromIndex(next_index_, samples_to_read, output->data_);
+  const size_t samples_read_per_channel = tot_samples_read / Channels();
+  next_index_ += samples_read_per_channel;
+  output->interleaved_ = true;
+  output->num_channels_ = Channels();
+  output->samples_per_channel_ = samples_read_per_channel;
 }
 
 void SyncBuffer::IncreaseEndTimestamp(uint32_t increment) {
   end_timestamp_ += increment;
 }
 
 void SyncBuffer::Flush() {
   Zeros(Size());
   next_index_ = Size();
   end_timestamp_ = 0;
   dtmf_index_ = 0;
 }
 
 void SyncBuffer::set_next_index(size_t value) {
   // Cannot set |next_index_| larger than the size of the buffer.
   next_index_ = std::min(value, Size());
 }
 
 void SyncBuffer::set_dtmf_index(size_t value) {
   // Cannot set |dtmf_index_| larger than the size of the buffer.
   dtmf_index_ = std::min(value, Size());
 }
 
 }  // namespace webrtc