Move functionality out from AudioFrame and into AudioFrameOperations.

webrtc/modules/audio_coding/neteq/sync_buffer.cc

Index: webrtc/modules/audio_coding/neteq/sync_buffer.cc
diff --git a/webrtc/modules/audio_coding/neteq/sync_buffer.cc b/webrtc/modules/audio_coding/neteq/sync_buffer.cc
index f841f754a8568bdd2dcd349d3619048b4d0ca964..05e5547bb67ed6ae469c4c96c2b6d7076d5a0ec4 100644
--- a/webrtc/modules/audio_coding/neteq/sync_buffer.cc
+++ b/webrtc/modules/audio_coding/neteq/sync_buffer.cc
@@ -12,6 +12,7 @@
 
 #include "webrtc/base/checks.h"
 #include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
+#include "webrtc/modules/utility/include/audio_frame_operations.h"
 
 namespace webrtc {
 
@@ -74,7 +75,7 @@ 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();
+  AudioFrameOperations::Reset(output);
   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();