New statistics interface for APM

webrtc/media/engine/fakewebrtcvoiceengine.h

 /*
  *  Copyright (c) 2010 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.
  */
 
@@ skipping 33 matching lines @@
 
 #define WEBRTC_BOOL_STUB(method, args) \
   bool method args override { return true; }
 
 #define WEBRTC_BOOL_STUB_CONST(method, args) \
   bool method args const override { return true; }
 
 #define WEBRTC_VOID_STUB(method, args) \
   void method args override {}
 
+#define WEBRTC_VOID_STUB_CONST(method, args) \
+  void method args const override {}
+
 #define WEBRTC_FUNC(method, args) int method args override
 
 #define WEBRTC_VOID_FUNC(method, args) void method args override
 
 class FakeAudioProcessing : public webrtc::AudioProcessing {
  public:
   FakeAudioProcessing() : experimental_ns_enabled_(false) {}
 
   WEBRTC_STUB(Initialize, ())
   WEBRTC_STUB(Initialize, (
@@ skipping 49 matching lines @@
   WEBRTC_VOID_STUB(set_stream_key_pressed, (bool key_pressed));
   WEBRTC_VOID_STUB(set_delay_offset_ms, (int offset));
   WEBRTC_STUB_CONST(delay_offset_ms, ());
   WEBRTC_STUB(StartDebugRecording,
               (const char filename[kMaxFilenameSize], int64_t max_size_bytes));
   WEBRTC_STUB(StartDebugRecording, (FILE * handle, int64_t max_size_bytes));
   WEBRTC_STUB(StartDebugRecording, (FILE * handle));
   WEBRTC_STUB(StartDebugRecordingForPlatformFile, (rtc::PlatformFile handle));
   WEBRTC_STUB(StopDebugRecording, ());
   WEBRTC_VOID_STUB(UpdateHistogramsOnCallEnd, ());
+  WEBRTC_VOID_STUB(SetStatisticsEnabled, (bool enabled));
+  WEBRTC_VOID_STUB_CONST(GetStatistics,
+                         (webrtc::AudioProcessing::AudioProcessingStatistics *
+                          stats));
   webrtc::EchoCancellation* echo_cancellation() const override { return NULL; }
   webrtc::EchoControlMobile* echo_control_mobile() const override {
     return NULL;
   }
   webrtc::GainControl* gain_control() const override { return NULL; }
   webrtc::HighPassFilter* high_pass_filter() const override { return NULL; }
   webrtc::LevelEstimator* level_estimator() const override { return NULL; }
   webrtc::NoiseSuppression* noise_suppression() const override { return NULL; }
   webrtc::VoiceDetection* voice_detection() const override { return NULL; }
 
@@ skipping 428 matching lines @@
   webrtc::AecmModes aecm_mode_ = webrtc::kAecmSpeakerphone;
   webrtc::NsModes ns_mode_ = webrtc::kNsDefault;
   webrtc::AgcModes agc_mode_ = webrtc::kAgcDefault;
   webrtc::AgcConfig agc_config_;
   FakeAudioProcessing audio_processing_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_