Removed callback between old AudioConferenceMixer and OutputMixer. The audio frame with mixed audio…

webrtc/modules/audio_mixer/source/new_audio_conference_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 54 matching lines @@
   ~NewAudioConferenceMixerImpl();
 
   // Must be called after ctor.
   bool Init();
 
   // Module functions
   int64_t TimeUntilNextProcess() override;
   void Process() override;
 
   // NewAudioConferenceMixer functions
-  int32_t RegisterMixedStreamCallback(
-      OldAudioMixerOutputReceiver* mixReceiver) override;
-  int32_t UnRegisterMixedStreamCallback() override;
+  void Mix(AudioFrame*) override;

[tommi, 2016/07/06 19:42:17]

add param name

[aleloi, 2016/07/07 09:20:16]

Done.

[ivoc, 2016/07/07 13:46:16]

Try to keep the order of the functions the same as in the
new_audio_conference_mixer.h header.

[aleloi, 2016/07/07 14:29:12]

Done.

   int32_t SetMixabilityStatus(MixerAudioSource* participant,
                               bool mixable) override;
   bool MixabilityStatus(const MixerAudioSource& participant) const override;
   int32_t SetMinimumMixingFrequency(Frequency freq) override;
   int32_t SetAnonymousMixabilityStatus(MixerAudioSource* participant,
                                        bool mixable) override;
   bool AnonymousMixabilityStatus(
       const MixerAudioSource& participant) const override;
 
  private:
@@ skipping 56 matching lines @@
 
   bool LimitMixedAudio(AudioFrame* mixedAudio) const;
 
   std::unique_ptr<CriticalSectionWrapper> _crit;
   std::unique_ptr<CriticalSectionWrapper> _cbCrit;
 
   int32_t _id;
 
   Frequency _minimumMixingFreq;
 
-  // Mix result callback
-  OldAudioMixerOutputReceiver* _mixReceiver;
-
   // The current sample frequency and sample size when mixing.
   Frequency _outputFrequency;
   size_t _sampleSize;
 
   // Memory pool to avoid allocating/deallocating AudioFrames
   MemoryPool<AudioFrame>* _audioFramePool;
 
   // List of all participants. Note all lists are disjunct
   MixerAudioSourceList _participantList;  // May be mixed.
   // Always mixed, anonomously.
@@ skipping 12 matching lines @@
   // Counter keeping track of concurrent calls to process.
   // Note: should never be higher than 1 or lower than 0.
   int16_t _processCalls;
 
   // Used for inhibiting saturation in mixing.
   std::unique_ptr<AudioProcessing> _limiter;
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_MIXER_SOURCE_NEW_AUDIO_CONFERENCE_MIXER_IMPL_H_