Changed mixing api and removed resampler

webrtc/modules/audio_mixer/include/audio_mixer_defines.h

Index: webrtc/modules/audio_mixer/include/audio_mixer_defines.h
diff --git a/webrtc/modules/audio_mixer/include/audio_mixer_defines.h b/webrtc/modules/audio_mixer/include/audio_mixer_defines.h
index 253e8f788c0dbe20b0b7b7a1f671adbf62ca08f1..008f4d60f16279d247505cb46d32359c4f622fb0 100644
--- a/webrtc/modules/audio_mixer/include/audio_mixer_defines.h
+++ b/webrtc/modules/audio_mixer/include/audio_mixer_defines.h
@@ -46,10 +46,6 @@ class MixerAudioSource {
   // 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: