Move audio frame memory handling inside AudioMixer.

webrtc/audio/audio_receive_stream.cc

 /*
  *  Copyright (c) 2015 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 254 matching lines @@
     if (packet_time.timestamp >= 0)
       arrival_time_ms = (packet_time.timestamp + 500) / 1000;
     size_t payload_size = length - header.headerLength;
     remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_size,
                                               header);
   }
 
   return channel_proxy_->ReceivedRTPPacket(packet, length, packet_time);
 }
 
-AudioMixer::Source::AudioFrameWithInfo
-AudioReceiveStream::GetAudioFrameWithInfo(int sample_rate_hz) {
-  return channel_proxy_->GetAudioFrameWithInfo(sample_rate_hz);
+AudioMixer::Source::AudioFrameInfo AudioReceiveStream::GetAudioFrameWithInfo(
+    int sample_rate_hz,
+    AudioFrame* audio_frame) {
+  return channel_proxy_->GetAudioFrameWithInfo(sample_rate_hz, audio_frame);
 }
 
 int AudioReceiveStream::Ssrc() {
   return config_.rtp.local_ssrc;
 }
 
 VoiceEngine* AudioReceiveStream::voice_engine() const {
   internal::AudioState* audio_state =
       static_cast<internal::AudioState*>(audio_state_.get());
   VoiceEngine* voice_engine = audio_state->voice_engine();
   RTC_DCHECK(voice_engine);
   return voice_engine;
 }
 }  // namespace internal
 }  // namespace webrtc