Changed mixing api and removed resampler

webrtc/modules/audio_mixer/source/new_audio_conference_mixer_impl.h

 /*
  *  Copyright (c) 2012 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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_MIXER_SOURCE_NEW_AUDIO_CONFERENCE_MIXER_IMPL_H_
 #define WEBRTC_MODULES_AUDIO_MIXER_SOURCE_NEW_AUDIO_CONFERENCE_MIXER_IMPL_H_
 
 #include <list>
 #include <map>
 #include <memory>
 
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/engine_configurations.h"
 #include "webrtc/modules/audio_mixer/include/new_audio_conference_mixer.h"
 #include "webrtc/modules/audio_conference_mixer/source/memory_pool.h"
 #include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/common_audio/resampler/include/push_resampler.h"
 
 namespace webrtc {
 class AudioProcessing;
 class CriticalSectionWrapper;
 
 struct FrameAndMuteInfo {
   FrameAndMuteInfo(AudioFrame* f, bool m) : frame(f), muted(m) {}
   AudioFrame* frame;
   bool muted;
 };
@@ skipping 34 matching lines @@
 
   // Must be called after ctor.
   bool Init();
 
   // NewAudioConferenceMixer functions
   int32_t SetMixabilityStatus(MixerAudioSource* audio_source,
                               bool mixable) override;
   bool MixabilityStatus(const MixerAudioSource& audio_source) const override;
   int32_t SetAnonymousMixabilityStatus(MixerAudioSource* audio_source,
                                        bool mixable) override;
-  void Mix(AudioFrame* audio_frame_for_mixing) override;
+  void Mix(int sample_rate,
+           size_t number_of_channels,
+           void* audio_data) override;
   int32_t SetMinimumMixingFrequency(Frequency freq) override;
   bool AnonymousMixabilityStatus(
       const MixerAudioSource& audio_source) const override;
 
  private:
   // Set/get mix frequency
   int32_t SetOutputFrequency(const Frequency& frequency);
   Frequency OutputFrequency() const;
 
   // Compute what audio sources to mix from audio_source_list_. Ramp in
@@ skipping 61 matching lines @@
   // mixing.
   bool use_limiter_;
 
   uint32_t _timeStamp;
 
   // Ensures that Mix is called from the same thread.
   rtc::ThreadChecker thread_checker_;
 
   // Used for inhibiting saturation in mixing.
   std::unique_ptr<AudioProcessing> _limiter;
+  // Converts mixed audio to the audio device output rate.

[aleloi, 2016/08/05 09:52:22]

Moved here from the former 'OutputMixer'.

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

I thought the idea was that ACM/NetEq could deliver data to us already at the
output rate, so the mixer wouldn't need to do resampling?

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

Thank you. I just got reminded of that when working on connecting the mixer to
the channels. You are right. Resampling happens ACMReceiver::GetAudio. I will
remove in next patch set.

+  PushResampler<int16_t> resampler_;
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_MIXER_SOURCE_NEW_AUDIO_CONFERENCE_MIXER_IMPL_H_