Propagate muted info from VoE Channel to AudioConferenceMixer

webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h

 /*
  *  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 MixHistory;
 
 // A callback class that all mixer participants must inherit from/implement.
 class MixerParticipant
 {
 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) = 0;
+                                  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;
 
     MixHistory* _mixHistory;
 protected:
@@ skipping 11 matching lines @@
                                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_