Change NetEq::GetAudio to use AudioFrame

webrtc/modules/audio_coding/acm2/acm_receiver.h

Index: webrtc/modules/audio_coding/acm2/acm_receiver.h
diff --git a/webrtc/modules/audio_coding/acm2/acm_receiver.h b/webrtc/modules/audio_coding/acm2/acm_receiver.h
index f5ceb61fbdccdb41d27d804045160cd8787bad1a..8d64a3c1ea2f171a6c40dcfcc275f83c249e4df7 100644
--- a/webrtc/modules/audio_coding/acm2/acm_receiver.h
+++ b/webrtc/modules/audio_coding/acm2/acm_receiver.h
@@ -286,9 +286,6 @@ class AcmReceiver {
   const Decoder* last_audio_decoder_ GUARDED_BY(crit_sect_);
   AudioFrame::VADActivity previous_audio_activity_ GUARDED_BY(crit_sect_);
   ACMResampler resampler_ GUARDED_BY(crit_sect_);
-  // Used in GetAudio, declared as member to avoid allocating every 10ms.
-  // TODO(henrik.lundin) Stack-allocate in GetAudio instead?
-  std::unique_ptr<int16_t[]> audio_buffer_ GUARDED_BY(crit_sect_);
   std::unique_ptr<int16_t[]> last_audio_buffer_ GUARDED_BY(crit_sect_);
   CallStatistics call_stats_ GUARDED_BY(crit_sect_);
   NetEq* neteq_;