Move audio frame memory handling inside AudioMixer.

webrtc/modules/audio_mixer/audio_mixer_impl.h

 /*
  *  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.
  */
 
@@ skipping 17 matching lines @@
 typedef std::vector<AudioFrame*> AudioFrameList;
 
 class AudioMixerImpl : public AudioMixer {
  public:
   struct SourceStatus {
     SourceStatus(Source* audio_source, bool is_mixed, float gain)
         : audio_source(audio_source), is_mixed(is_mixed), gain(gain) {}
     Source* audio_source = nullptr;
     bool is_mixed = false;
     float gain = 0.0f;
+
+    // A frame that will be passed to audio_source->GetAudioFrameWithInfo.
+    AudioFrame audio_frame;
   };
 
-  typedef std::vector<SourceStatus> SourceStatusList;
+  using SourceStatusList = std::vector<std::unique_ptr<SourceStatus>>;
 
   // AudioProcessing only accepts 10 ms frames.
   static const int kFrameDurationInMs = 10;
   static const int kMaximumAmountOfMixedAudioSources = 3;
   static const int kDefaultFrequency = 48000;
 
   static rtc::scoped_refptr<AudioMixerImpl> Create();
 
   ~AudioMixerImpl() override;
 
@@ skipping 52 matching lines @@
   rtc::ThreadChecker thread_checker_;
 
   // Used for inhibiting saturation in mixing.
   std::unique_ptr<AudioProcessing> limiter_ ACCESS_ON(&thread_checker_);
 
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioMixerImpl);
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_MIXER_AUDIO_MIXER_IMPL_H_