Removed the memory pool from the mixer.

webrtc/modules/audio_mixer/include/audio_mixer_defines.h

 /*
  *  Copyright (c) 2011 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 27 matching lines @@
   // with new audio every time it's called. The return value will be
   // interpreted as follows.
   enum class AudioFrameInfo {
     kNormal,  // The samples in audio_frame are valid and should be used.
     kMuted,   // The samples in audio_frame should not be used, but should be
               // implicitly interpreted as zero. Other fields in audio_frame
               // may be read and should contain meaningful values.
     kError    // audio_frame will not be used.
   };
 
   virtual AudioFrameInfo GetAudioFrameWithMuted(int32_t id,

[ossu, 2016/07/06 11:49:37]

So, why does this function need an AudioFrame as input if it's going to return
one of its own anyways?

[aleloi, 2016/07/06 11:59:38]

It's public API (although only defined in voe::Channel, one test and something
called Buzz). I could change it, but then I would have to change the code
everywhere else as well.

[ossu, 2016/07/06 12:58:44]

Well it's not public yet, since you just forked the old mixer code and started
on your own. :)

                                                 AudioFrame* audio_frame) {
     return GetAudioFrame(id, audio_frame) == -1 ? AudioFrameInfo::kError
                                                 : AudioFrameInfo::kNormal;
   }
 
   // Returns true if the participant was mixed this mix iteration.
   bool IsMixed() const;
 
+  // Indended use: the caller of GetAudioFrame* uses this method to
+  // provide an AudioFrame to this object. This decreases complexity
+  // and the need to allocate AudioFrames on the caller side.
+  AudioFrame* GetFramePointer() { return &_audio_frame; }
+
   // This function specifies the sampling frequency needed for the AudioFrame
   // for future GetAudioFrame(..) calls.
   virtual int32_t NeededFrequency(int32_t id) const = 0;
 
   NewMixHistory* _mixHistory;
 
  protected:
   MixerAudioSource();
   virtual ~MixerAudioSource();
+
+ private:
+  // This frame is intended to be passed to GetAudioFrame* for
+  // efficiency.
+  AudioFrame _audio_frame;

[ossu, 2016/07/06 11:49:37]

Seems to me the frame should be put in the implementing class, not in this one.
Also it should be spelled audio_frame_. :)

[aleloi, 2016/07/06 11:59:38]

You are right. I think I wanted to be consistent with the surrounding code or
something.

 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_MIXER_INCLUDE_AUDIO_MIXER_DEFINES_H_