Changed ramping functionality of the AudioMixer.

webrtc/modules/audio_mixer/audio_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_AUDIO_MIXER_IMPL_H_
 #define WEBRTC_MODULES_AUDIO_MIXER_AUDIO_MIXER_IMPL_H_
 
 #include <map>
 #include <memory>
 #include <vector>
 
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/modules/audio_mixer/audio_mixer.h"
-#include "webrtc/modules/audio_mixer/audio_source_with_mix_status.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/voice_engine/level_indicator.h"
 #include "webrtc/voice_engine_configurations.h"
 
 namespace webrtc {
 
 typedef std::vector<AudioFrame*> AudioFrameList;
-typedef std::vector<AudioSourceWithMixStatus> MixerAudioSourceList;
 
 class AudioMixerImpl : public AudioMixer {
  public:
+  struct SourceStatus {
+    SourceStatus(Source* audio_source, bool is_mixed, float gain)
+        : audio_source_(audio_source), is_mixed_(is_mixed), gain_(gain) {}
+    Source* audio_source_ = nullptr;

[hlundin-webrtc, 2016/10/07 15:03:10]

Structs don't have trailing underscore on the member names.
https://google.github.io/styleguide/cppguide.html#Variable_Names

[aleloi, 2016/10/10 10:58:17]

Thanks! I think I should re-read the style guide again.

+    bool is_mixed_ = false;
+    float gain_ = 0;
+  };
+
+  typedef std::vector<SourceStatus> MixerAudioSourceList;

[hlundin-webrtc, 2016/10/07 15:03:10]

How about SourceStatusList?

[aleloi, 2016/10/10 10:58:17]

Done.

+
   // AudioProcessing only accepts 10 ms frames.
   static const int kFrameDurationInMs = 10;
 
   static std::unique_ptr<AudioMixerImpl> Create(int id);
 
   ~AudioMixerImpl() override;
 
   // AudioMixer functions
   int32_t SetMixabilityStatus(Source* audio_source, bool mixable) override;
   bool MixabilityStatus(const Source& audio_source) const override;
@@ skipping 67 matching lines @@
   std::unique_ptr<AudioProcessing> limiter_ ACCESS_ON(&thread_checker_);
 
   // Measures audio level for the combined signal.
   voe::AudioLevel audio_level_ ACCESS_ON(&thread_checker_);
 
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioMixerImpl);
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_MIXER_AUDIO_MIXER_IMPL_H_