Changed mixing api and removed resampler

webrtc/modules/audio_mixer/audio_mixer.h

Index: webrtc/modules/audio_mixer/audio_mixer.h
diff --git a/webrtc/modules/audio_mixer/audio_mixer.h b/webrtc/modules/audio_mixer/audio_mixer.h
index c50a2cfa708dbf394b6be3b90c284573c41ad6c7..9aabfcca5b751a885e93aa67e11edef5f2ae805d 100644
--- a/webrtc/modules/audio_mixer/audio_mixer.h
+++ b/webrtc/modules/audio_mixer/audio_mixer.h
@@ -48,8 +48,6 @@ class AudioMixer : public FileCallback {
 
   int DeRegisterExternalMediaProcessing();
 
-  int32_t MixActiveChannels();

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

This file is the former 'OutputMixer'. Parts of its contents are slowly being
merged into the former 'AudioConferenceMixer' to create a single module
responsible for mixing. It is not deleted yet because there are several tests
that would have to be removed, and because not all content is yet in
'NewAudioConferenceMixer'.

In this CL, the resampling is moved to the old mixer. As a consequence, some
APIs have to change. This function asked the conference mixer to mix, but it
cannot do that anymore, because it doesn't have the sample rate and channel
count.

-
   int32_t DoOperationsOnCombinedSignal(bool feed_data_to_apm);
 
   int32_t SetMixabilityStatus(MixerAudioSource& audio_source,  // NOLINT
@@ -100,8 +98,6 @@ class AudioMixer : public FileCallback {
   rtc::CriticalSection _fileCritSect;
   NewAudioConferenceMixer& _mixerModule;
   AudioFrame _audioFrame;
-  // Converts mixed audio to the audio device output rate.

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

Moved to NewAudioConferenceMixer.

-  PushResampler<int16_t> resampler_;
   // Converts mixed audio to the audio processing rate.
   PushResampler<int16_t> audioproc_resampler_;
   AudioLevel _audioLevel;  // measures audio level for the combined signal