Changed mixing api and removed resampler

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 28 matching lines @@
   // classes are allowed to return the same AudioFrame pointer on
   // different calls. The pointer must stay valid until the next
   // mixing call or until this audio source is disconnected from the
   // mixer.
   virtual AudioFrameWithMuted GetAudioFrameWithMuted(int32_t id,
                                                      int sample_rate_hz) = 0;
 
   // Returns true if the participant was mixed this mix iteration.
   bool IsMixed() const;
 
-  // This function specifies the sampling frequency needed for the AudioFrame
-  // for future GetAudioFrame(..) calls.
-  virtual int32_t NeededFrequency(int32_t id) const = 0;
-

[aleloi2, 2016/08/06 10:13:49]

I removed NeededFrequency, since the mixer asks participants for data at a
specific rate anyway. This would fix a TODO(andrew) in
AudioMixer::GetMixedAudio. The mixer now mixes at the rate asked for by the
audio device module. Channel::NeededFrequency was computed from the playout and
receiving frequency, so there shouldn't be much (or any) performance
differences, just simplifications.

   NewMixHistory* _mixHistory;
 
  protected:
   MixerAudioSource();
   virtual ~MixerAudioSource();
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_MIXER_INCLUDE_AUDIO_MIXER_DEFINES_H_