Move audio frame memory handling inside AudioMixer.

webrtc/modules/audio_mixer/audio_mixer_impl.h

Index: webrtc/modules/audio_mixer/audio_mixer_impl.h
diff --git a/webrtc/modules/audio_mixer/audio_mixer_impl.h b/webrtc/modules/audio_mixer/audio_mixer_impl.h
index 500bb7862e5defe2b9fd906c057c6cf60fec3087..1abc679480e69d35920d39eb1bb68693c622e1ac 100644
--- a/webrtc/modules/audio_mixer/audio_mixer_impl.h
+++ b/webrtc/modules/audio_mixer/audio_mixer_impl.h
@@ -35,9 +35,12 @@ class AudioMixerImpl : public AudioMixer {
     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;