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
-  // called from a single thread.
-  virtual void Mix(AudioFrame* audio_frame_for_mixing) = 0;

[aleloi, 2016/08/05 09:52:21]

We are moving away from the AudioFrame. In the future, we want to communicate
audio with simpler structures. The 'void*' is a step in that direction. We also
may want to mix floating point audio.

This will we be main mixer function for clients of the mixer. The mixer will do
necessary resampling.

[the sun, 2016/08/05 12:26:00]

Then why not float* instead of void*?

[aleloi, 2016/08/05 12:37:14]

To do one change at a time, probably. Much of the mixer and surrounding code
needs to be changed to mix in floats, I think. In what order do you think it's
simpler to do the changes: rewrite mixer and surroundings to mix in floats, or
plug in the new mixer? Curious.

+  // mixed result is placed in the provided buffer, which is
+  // guaranteed to be large enough. Can only be called from a single
+  // thread.
+  virtual void Mix(int sample_rate,
+                   size_t number_of_channels,
+                   void* audio_data) = 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;
 
   // 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_