Changed mixing api and removed resampler

webrtc/modules/audio_mixer/include/new_audio_conference_mixer.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 30 matching lines @@
   virtual bool MixabilityStatus(const MixerAudioSource& audio_source) const = 0;
 
   // Inform the mixer that the audio source should always be mixed and not
   // count toward the number of mixed audio sources. Note that an audio source
   // must have been added to the mixer (by calling SetMixabilityStatus())
   // before this function can be successfully called.
   virtual int32_t SetAnonymousMixabilityStatus(MixerAudioSource* audio_source,
                                                bool mixable) = 0;
 
   // Performs mixing by asking registered audio sources for audio. The
   // mixed result is placed in the provided AudioFrame. Can only be

[minyue-webrtc, 2016/08/08 15:05:44]

add comments on the new arguments

   // called from a single thread.
-  virtual void Mix(AudioFrame* audio_frame_for_mixing) = 0;
-
-  // Set the minimum sampling frequency at which to mix. The mixing algorithm
-  // may still choose to mix at a higher samling frequency to avoid
-  // downsampling of audio contributing to the mixed audio.
-  virtual int32_t SetMinimumMixingFrequency(Frequency freq) = 0;
+  virtual void Mix(int sample_rate,
+                   size_t number_of_channels,
+                   AudioFrame* audio_frame_for_mixing) = 0;
 
   // Returns true if the audio source is mixed anonymously.
   virtual bool AnonymousMixabilityStatus(
       const MixerAudioSource& audio_source) const = 0;
 
  protected:
   NewAudioConferenceMixer() {}
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_MIXER_INCLUDE_NEW_AUDIO_CONFERENCE_MIXER_H_