Changed mixing api and removed resampler

webrtc/modules/audio_mixer/include/new_audio_conference_mixer.h

Index: webrtc/modules/audio_mixer/include/new_audio_conference_mixer.h
diff --git a/webrtc/modules/audio_mixer/include/new_audio_conference_mixer.h b/webrtc/modules/audio_mixer/include/new_audio_conference_mixer.h
index 844e9ae0fdb67d7d209f141c33e891370d750148..a5b718b8a2e524a7303ae3253d5fc465022d531e 100644
--- a/webrtc/modules/audio_mixer/include/new_audio_conference_mixer.h
+++ b/webrtc/modules/audio_mixer/include/new_audio_conference_mixer.h
@@ -48,9 +48,12 @@ class NewAudioConferenceMixer {
                                                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;
+  // 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,
+                   AudioFrame* audio_frame_for_mixing) = 0;

[aleloi2, 2016/08/06 10:13:50]

I decided to leave the AudioFrame in the mixer for this CL. Replacing it with
int16_t* would require to reimplement the functions in AudioFrame and
AudioFrameOperations (which I don't have a concrete plan for yet). It's shifting
(>>=), adding, remixing (MonoToStereo, StereoToMono),
audio_processing::ProcessReverseStream, AudioLevel::ComputeLevel, and probably a
few more. The receiver side is simpler, since data is mostly just copied to the
audio device buffer in VoE::GetPlayoutData.

 
   // Set the minimum sampling frequency at which to mix. The mixing algorithm
   // may still choose to mix at a higher samling frequency to avoid