Added empty directory with myself as owner for new mixer.

webrtc/modules/audio_mixer/include/audio_conference_mixer_defines.h

Index: webrtc/modules/audio_mixer/include/audio_conference_mixer_defines.h
diff --git a/webrtc/modules/audio_mixer/include/audio_conference_mixer_defines.h b/webrtc/modules/audio_mixer/include/audio_conference_mixer_defines.h
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index 0000000000000000000000000000000000000000..d1feb98939d655022b7ea68c7eff354ee083506b
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+++ b/webrtc/modules/audio_mixer/include/audio_conference_mixer_defines.h
@@ -0,0 +1,84 @@
+/*
+ *  Copyright (c) 2011 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_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
+#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
+
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+class NewMixHistory;
+
+// A callback class that all mixer participants must inherit from/implement.
+class MixerAudioSource {
+ public:
+  // The implementation of this function should update audioFrame with new
+  // audio every time it's called.
+  //
+  // If it returns -1, the frame will not be added to the mix.
+  //
+  // NOTE: This function should not be called. It will remain for a short
+  // time so that subclasses can override it without getting warnings.
+  // TODO(henrik.lundin) Remove this function.
+  virtual int32_t GetAudioFrame(int32_t id, AudioFrame* audioFrame) {
+    RTC_CHECK(false);
+    return -1;
+  }
+
+  // The implementation of GetAudioFrameWithMuted should update audio_frame
+  // with new audio every time it's called. The return value will be
+  // interpreted as follows.
+  enum class AudioFrameInfo {
+    kNormal,  // The samples in audio_frame are valid and should be used.
+    kMuted,   // The samples in audio_frame should not be used, but should be
+              // implicitly interpreted as zero. Other fields in audio_frame
+              // may be read and should contain meaningful values.
+    kError    // audio_frame will not be used.
+  };
+
+  virtual AudioFrameInfo GetAudioFrameWithMuted(int32_t id,
+                                                AudioFrame* audio_frame) {
+    return GetAudioFrame(id, audio_frame) == -1 ? AudioFrameInfo::kError
+                                                : AudioFrameInfo::kNormal;
+  }
+
+  // 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;
+
+  NewMixHistory* _mixHistory;
+
+ protected:
+  MixerAudioSource();
+  virtual ~MixerAudioSource();
+};
+
+class AudioMixerOutputReceiver {
+ public:
+  // This callback function provides the mixed audio for this mix iteration.
+  // Note that uniqueAudioFrames is an array of AudioFrame pointers with the
+  // size according to the size parameter.
+  virtual void NewMixedAudio(const int32_t id,
+                             const AudioFrame& generalAudioFrame,
+                             const AudioFrame** uniqueAudioFrames,
+                             const uint32_t size) = 0;
+
+ protected:
+  AudioMixerOutputReceiver() {}
+  virtual ~AudioMixerOutputReceiver() {}
+};
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_DEFINES_H_