webrtc_payoda_rms.patch

 talk/app/webrtc/audiotrack.cc                      | 13 ++++++++++--
 talk/app/webrtc/audiotrack.h                       |  9 +++++++++
 talk/app/webrtc/mediastreamhandler.cc              |  5 +++++
 talk/app/webrtc/mediastreaminterface.h             |  9 +++++++++
 talk/app/webrtc/mediastreamprovider.h              |  6 ++++++
 talk/app/webrtc/mediastreamtrackproxy.h            |  6 ++++++
 talk/app/webrtc/statscollector_unittest.cc         |  4 ++++
 talk/app/webrtc/webrtcsession.cc                   | 23 ++++++++++++++++++++++
 talk/app/webrtc/webrtcsession.h                    |  7 +++++++
 talk/media/base/mediaengine.h                      |  9 +++++++++
 talk/media/webrtc/webrtcmediaengine.cc             | 11 +++++++++++
 talk/media/webrtc/webrtcvoiceengine.cc             | 21 ++++++++++++++++++++
 talk/media/webrtc/webrtcvoiceengine.h              |  7 +++++++
 talk/session/media/channel.cc                      | 20 +++++++++++++++++++
 talk/session/media/channel.h                       |  9 +++++++++
 talk/session/media/channelmanager_unittest.cc      |  1 +
 webrtc/base/win32window.cc                         | 14 ++++++++++---
 .../neteq/neteq_network_stats_unittest.cc          |  3 ++-
 webrtc/modules/audio_processing/rms_level.cc       |  6 ++++--
 webrtc/modules/audio_processing/rms_level.h        |  2 +-
 webrtc/modules/interface/module_common_types.h     |  9 +++++++++
 .../video_capture/windows/video_capture_ds.cc      |  3 ++-
 webrtc/voice_engine/channel.cc                     | 11 +++++++++--
 webrtc/voice_engine/channel.h                      |  3 +++
 webrtc/voice_engine/include/voe_external_media.h   |  5 +++++
 webrtc/voice_engine/shared_data.cc                 |  2 ++
 webrtc/voice_engine/shared_data.h                  |  8 ++++++++
 webrtc/voice_engine/transmit_mixer.cc              |  6 ++++--
 webrtc/voice_engine/transmit_mixer.h               |  3 +++
 webrtc/voice_engine/voe_base_impl.cc               |  7 +++++++
 webrtc/voice_engine/voice_engine_impl.cc           | 11 +++++++++++
 webrtc/voice_engine/voice_engine_impl.h            |  6 ++++++
 32 files changed, 245 insertions(+), 14 deletions(-)

diff --git a/talk/app/webrtc/audiotrack.cc b/talk/app/webrtc/audiotrack.cc
index b0c9129..647a2c0 100644
--- a/talk/app/webrtc/audiotrack.cc
+++ b/talk/app/webrtc/audiotrack.cc
@@ -34,11 +34,20 @@ namespace webrtc {
 static const char kAudioTrackKind[] = "audio";
 
 AudioTrack::AudioTrack(const std::string& label,
-                       AudioSourceInterface* audio_source)
-    : MediaStreamTrack<AudioTrackInterface>(label),
+	AudioSourceInterface* audio_source)
+	: MediaStreamTrack<AudioTrackInterface>(label),
+	rms_(NULL),
       audio_source_(audio_source) {
 }
 
+// TODO(dmi): Added by Doubango Telecom
+void AudioTrack::AddRMS(AudioRMSInterface* rms) {
+	rms_ = rms;
+}
+void AudioTrack::RemoveRMS(AudioRMSInterface* rms) {
+	rms_ = NULL;
+}
+
 std::string AudioTrack::kind() const {
   return kAudioTrackKind;
 }
diff --git a/talk/app/webrtc/audiotrack.h b/talk/app/webrtc/audiotrack.h
index 750f272..718e89a 100644
--- a/talk/app/webrtc/audiotrack.h
+++ b/talk/app/webrtc/audiotrack.h
@@ -36,6 +36,8 @@
 
 namespace webrtc {
 
+	class AudioRMSInterface;
+
 class AudioTrack : public MediaStreamTrack<AudioTrackInterface> {
  public:
   static rtc::scoped_refptr<AudioTrack> Create(
@@ -45,6 +47,12 @@ class AudioTrack : public MediaStreamTrack<AudioTrackInterface> {
   AudioSourceInterface* GetSource() const override {
     return audio_source_.get();
   }
+
+  // TODO(dmi): Added by Doubango Telecom
+  virtual void AddRMS(AudioRMSInterface* rms)override;
+  virtual void RemoveRMS(AudioRMSInterface* rms)override;
+  virtual AudioRMSInterface* GetRMS()const override { return rms_; }
+
   // TODO(xians): Implement these methods.
   void AddSink(AudioTrackSinkInterface* sink) override {}
   void RemoveSink(AudioTrackSinkInterface* sink) override {}
@@ -62,6 +70,7 @@ class AudioTrack : public MediaStreamTrack<AudioTrackInterface> {
 
  private:
   rtc::scoped_refptr<AudioSourceInterface> audio_source_;
+  rtc::scoped_refptr<AudioRMSInterface> rms_;
 };
 
 }  // namespace webrtc
diff --git a/talk/app/webrtc/mediastreamhandler.cc b/talk/app/webrtc/mediastreamhandler.cc
index f68699d..1b7fec2 100644
--- a/talk/app/webrtc/mediastreamhandler.cc
+++ b/talk/app/webrtc/mediastreamhandler.cc
@@ -92,6 +92,8 @@ LocalAudioTrackHandler::LocalAudioTrackHandler(
       sink_adapter_(new LocalAudioSinkAdapter()) {
   OnEnabledChanged();
   track->AddSink(sink_adapter_.get());
+  // TODO(dmi): added by Doubango Telecom
+  provider_->AddRMS(audio_track_->GetRMS());
 }
 
 LocalAudioTrackHandler::~LocalAudioTrackHandler() {
@@ -116,6 +118,9 @@ void LocalAudioTrackHandler::OnEnabledChanged() {
         audio_track_->GetSource())->options();
   }
 
+  // TODO(dmi): added by Doubango Telecom
+  provider_->AddRMS(audio_track_->GetRMS());
+
   // Use the renderer if the audio track has one, otherwise use the sink
   // adapter owned by this class.
   cricket::AudioRenderer* renderer = audio_track_->GetRenderer() ?
diff --git a/talk/app/webrtc/mediastreaminterface.h b/talk/app/webrtc/mediastreaminterface.h
index 4de0648..d69ee7a 100644
--- a/talk/app/webrtc/mediastreaminterface.h
+++ b/talk/app/webrtc/mediastreaminterface.h
@@ -40,6 +40,7 @@
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/refcount.h"
 #include "webrtc/base/scoped_ref_ptr.h"
+#include "webrtc/modules/interface/module_common_types.h"
 
 namespace cricket {
 
@@ -175,6 +176,9 @@ class AudioSourceInterface : public MediaSourceInterface {
   // Sets the volume to the source. |volume| is in  the range of [0, 10].
   virtual void SetVolume(double volume) {}
 
+  // TODO(dmi): Added by Doubango Telecom
+  virtual bool GetSignalLevel(int* level) { return false; }
+
   // Registers/unregisters observer to the audio source.
   virtual void RegisterAudioObserver(AudioObserver* observer) {}
   virtual void UnregisterAudioObserver(AudioObserver* observer) {}
@@ -229,6 +233,11 @@ class AudioTrackInterface : public MediaStreamTrackInterface {
   virtual void AddSink(AudioTrackSinkInterface* sink) = 0;
   virtual void RemoveSink(AudioTrackSinkInterface* sink) = 0;
 
+  // TODO(dmi): Added by Doubango Telecom
+  virtual void AddRMS(AudioRMSInterface* rms) {};
+  virtual void RemoveRMS(AudioRMSInterface* rms) {};
+  virtual AudioRMSInterface* GetRMS()const { return NULL; };
+
   // Get the signal level from the audio track.
   // Return true on success, otherwise false.
   // TODO(xians): Change the interface to int GetSignalLevel() and pure virtual
diff --git a/talk/app/webrtc/mediastreamprovider.h b/talk/app/webrtc/mediastreamprovider.h
index e903fda..e006539 100644
--- a/talk/app/webrtc/mediastreamprovider.h
+++ b/talk/app/webrtc/mediastreamprovider.h
@@ -39,6 +39,7 @@ struct VideoOptions;
 }  // namespace cricket
 
 namespace webrtc {
+	class AudioRMSInterface;
 
 // This interface is called by AudioTrackHandler classes in mediastreamhandler.h
 // to change the settings of an audio track connected to certain PeerConnection.
@@ -57,6 +58,11 @@ class AudioProviderInterface {
   // |volume| is in the range of [0, 10].
   virtual void SetAudioPlayoutVolume(uint32 ssrc, double volume) = 0;
 
+  // TODO(dmi): Added by Doubango Telecom
+  virtual void AddRMS(AudioRMSInterface* rms) {};
+  virtual void RemoveRMS(AudioRMSInterface* rms) {};
+  virtual AudioRMSInterface* GetRMS()const { return NULL; };
+
  protected:
   virtual ~AudioProviderInterface() {}
 };
diff --git a/talk/app/webrtc/mediastreamtrackproxy.h b/talk/app/webrtc/mediastreamtrackproxy.h
index d97ba3c..2bd7452 100644
--- a/talk/app/webrtc/mediastreamtrackproxy.h
+++ b/talk/app/webrtc/mediastreamtrackproxy.h
@@ -45,6 +45,12 @@ BEGIN_PROXY_MAP(AudioTrack)
   PROXY_METHOD1(void, AddSink, AudioTrackSinkInterface*)
   PROXY_METHOD1(void, RemoveSink, AudioTrackSinkInterface*)
   PROXY_METHOD1(bool, GetSignalLevel, int*)
+
+  // TODO(dmi): Added by Doubango Telecom
+  PROXY_METHOD1(void, AddRMS, AudioRMSInterface*)
+  PROXY_METHOD1(void, RemoveRMS, AudioRMSInterface*)
+  PROXY_CONSTMETHOD0(AudioRMSInterface*, GetRMS)
+
   PROXY_METHOD0(rtc::scoped_refptr<AudioProcessorInterface>,
                 GetAudioProcessor)
   PROXY_METHOD0(cricket::AudioRenderer*, GetRenderer)
diff --git a/talk/app/webrtc/statscollector_unittest.cc b/talk/app/webrtc/statscollector_unittest.cc
index ab9181f..e36e7c4 100644
--- a/talk/app/webrtc/statscollector_unittest.cc
+++ b/talk/app/webrtc/statscollector_unittest.cc
@@ -137,6 +137,10 @@ class FakeAudioTrack
     *level = 1;
     return true;
   }
+  // TODO(dmi): added by Doubango Telecom
+  void AddRMS(AudioRMSInterface* rms)override {}
+  void RemoveRMS(AudioRMSInterface* rms)override {}
+  virtual AudioRMSInterface* GetRMS()const override { return NULL; }
   rtc::scoped_refptr<webrtc::AudioProcessorInterface> GetAudioProcessor()
       override {
     return processor_;
diff --git a/talk/app/webrtc/webrtcsession.cc b/talk/app/webrtc/webrtcsession.cc
index b6a178c..0af3c77 100644
--- a/talk/app/webrtc/webrtcsession.cc
+++ b/talk/app/webrtc/webrtcsession.cc
@@ -495,6 +495,7 @@ WebRtcSession::WebRtcSession(
       dtls_enabled_(false),
       data_channel_type_(cricket::DCT_NONE),
       ice_restart_latch_(new IceRestartAnswerLatch),
+	  rms_(NULL),
       metrics_observer_(NULL) {
 }
 
@@ -507,6 +508,7 @@ WebRtcSession::~WebRtcSession() {
     channel_manager_->DestroyVideoChannel(video_channel_.release());
   }
   if (voice_channel_) {
+	voice_channel_->RemoveRMS(rms_);
     SignalVoiceChannelDestroyed();
     channel_manager_->DestroyVoiceChannel(voice_channel_.release(), nullptr);
   }
@@ -517,6 +519,7 @@ WebRtcSession::~WebRtcSession() {
   for (size_t i = 0; i < saved_candidates_.size(); ++i) {
     delete saved_candidates_[i];
   }
+  rms_ = NULL;
   delete identity();
 }
 
@@ -1159,6 +1162,26 @@ void WebRtcSession::SetVideoSend(uint32 ssrc, bool enable,
     video_channel_->SetChannelOptions(*options);
 }
 
+// TODO(dmi): Added by Doubango Telecom
+void WebRtcSession::AddRMS(AudioRMSInterface* rms) {
+	ASSERT(signaling_thread()->IsCurrent());
+	rms_ = rms;
+	if (!voice_channel_) {
+		LOG(LS_ERROR) << "AddRMS: No audio channel exists.";
+		return;
+	}
+	voice_channel_->AddRMS(rms_);
+}
+void WebRtcSession::RemoveRMS(AudioRMSInterface* rms) {
+	rms_ = NULL;
+	if (voice_channel_) {
+		voice_channel_->RemoveRMS(rms);
+	}
+}
+AudioRMSInterface* WebRtcSession::GetRMS()const {
+	return rms_;
+}
+
 bool WebRtcSession::CanInsertDtmf(const std::string& track_id) {
   ASSERT(signaling_thread()->IsCurrent());
   if (!voice_channel_) {
diff --git a/talk/app/webrtc/webrtcsession.h b/talk/app/webrtc/webrtcsession.h
index 30ebc1e..56c376c 100644
--- a/talk/app/webrtc/webrtcsession.h
+++ b/talk/app/webrtc/webrtcsession.h
@@ -204,6 +204,10 @@ class WebRtcSession : public cricket::BaseSession,
                     const cricket::AudioOptions& options,
                     cricket::AudioRenderer* renderer) override;
   void SetAudioPlayoutVolume(uint32 ssrc, double volume) override;
+  // TODO(dmi): Added by Doubango Telecom
+  virtual void AddRMS(AudioRMSInterface* rms)override;
+  virtual void RemoveRMS(AudioRMSInterface* rms)override;
+  virtual AudioRMSInterface* GetRMS()const override;
 
   // Implements VideoMediaProviderInterface.
   bool SetCaptureDevice(uint32 ssrc, cricket::VideoCapturer* camera) override;
@@ -411,6 +415,9 @@ class WebRtcSession : public cricket::BaseSession,
   // Declares the RTCP mux policy for the WebRTCSession.
   PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy_;
 
+  // TODO(dmi): Doubango Telcom
+  rtc::scoped_refptr<AudioRMSInterface> rms_;
+
   DISALLOW_COPY_AND_ASSIGN(WebRtcSession);
 };
 }  // namespace webrtc
diff --git a/talk/media/base/mediaengine.h b/talk/media/base/mediaengine.h
index 501333f..0f693c7 100644
--- a/talk/media/base/mediaengine.h
+++ b/talk/media/base/mediaengine.h
@@ -52,6 +52,10 @@
 #define DISABLE_MEDIA_ENGINE_FACTORY
 #endif
 
+namespace webrtc {
+	class AudioRMSInterface;
+}
+
 namespace cricket {
 
 class VideoCapturer;
@@ -137,6 +141,11 @@ class MediaEngineInterface {
   virtual bool UnregisterVoiceProcessor(uint32 ssrc,
                                         VoiceProcessor* video_processor,
                                         MediaProcessorDirection direction) = 0;
+
+  // TODO(dmi): Added by Doubango Telecom
+  virtual void AddRMS(webrtc::AudioRMSInterface* rms) {}
+  virtual void RemoveRMS(webrtc::AudioRMSInterface* rms) {}
+  virtual webrtc::AudioRMSInterface* GetRMS()const { return NULL; }
 };
 
 
diff --git a/talk/media/webrtc/webrtcmediaengine.cc b/talk/media/webrtc/webrtcmediaengine.cc
index cf6a5cf..9081acd 100644
--- a/talk/media/webrtc/webrtcmediaengine.cc
+++ b/talk/media/webrtc/webrtcmediaengine.cc
@@ -41,6 +41,17 @@ class WebRtcMediaEngine2
     video_.SetExternalDecoderFactory(decoder_factory);
     video_.SetExternalEncoderFactory(encoder_factory);
   }
+  // TODO(dmi): Added by Doubango Telecom
+  virtual void AddRMS(webrtc::AudioRMSInterface* rms) {
+	  voice_.AddRMS(rms);
+  }
+  virtual void RemoveRMS(webrtc::AudioRMSInterface* rms) {
+	  voice_.RemoveRMS(rms);
+  }
+  virtual webrtc::AudioRMSInterface* GetRMS()const {
+	  return voice_.GetRMS();
+  }
+private:
 };
 
 }  // namespace cricket
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
index e288e70..bfcb280 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -1617,6 +1617,27 @@ int WebRtcVoiceEngine::CreateMediaVoiceChannel() {
   return CreateVoiceChannel(voe_wrapper_.get());
 }
 
+// TODO(dmi): Doubango Telecom
+void WebRtcVoiceEngine::AddRMS(webrtc::AudioRMSInterface* rms) {
+	ASSERT(voe_wrapper_ != NULL);
+	if (voe_wrapper_) {
+		voe_wrapper_->media()->AddRMS(rms);
+	}
+}
+void WebRtcVoiceEngine::RemoveRMS(webrtc::AudioRMSInterface* rms) {
+	ASSERT(voe_wrapper_ != NULL);
+	if (voe_wrapper_) {
+		voe_wrapper_->media()->RemoveRMS(rms);
+	}
+}
+webrtc::AudioRMSInterface* WebRtcVoiceEngine::GetRMS()const {
+	ASSERT(voe_wrapper_ != NULL);
+	if (voe_wrapper_) {
+		return voe_wrapper_->media()->GetRMS();
+	}
+	return NULL;
+}
+
 class WebRtcVoiceMediaChannel::WebRtcVoiceChannelRenderer
     : public AudioRenderer::Sink {
  public:
diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h
index c8e7980..465f11b 100644
--- a/talk/media/webrtc/webrtcvoiceengine.h
+++ b/talk/media/webrtc/webrtcvoiceengine.h
@@ -49,6 +49,7 @@
 
 namespace webrtc {
 class VideoEngine;
+class AudioRMSInterface;
 }
 
 namespace cricket {
@@ -159,6 +160,12 @@ class WebRtcVoiceEngine
   // Create a VoiceEngine Channel.
   int CreateMediaVoiceChannel();
 
+  // TODO(dmi): Added by Doubango Telecom
+  virtual void AddRMS(webrtc::AudioRMSInterface* rms);
+  virtual void RemoveRMS(webrtc::AudioRMSInterface* rms);
+  virtual webrtc::AudioRMSInterface* GetRMS()const;
+
+
  private:
   typedef std::vector<WebRtcVoiceMediaChannel *> ChannelList;
   typedef sigslot::
diff --git a/talk/session/media/channel.cc b/talk/session/media/channel.cc
index 8de8d37..b8b4ee7 100644
--- a/talk/session/media/channel.cc
+++ b/talk/session/media/channel.cc
@@ -1587,6 +1587,26 @@ bool VoiceChannel::SetChannelOptions(const AudioOptions& options) {
                              media_channel(), options));
 }
 
+// TODO(dmi): Added by Doubango Telecom
+void VoiceChannel::AddRMS(webrtc::AudioRMSInterface* rms) {
+	ASSERT(media_engine_ != NULL);
+	if (media_engine_) {
+		media_engine_->AddRMS(rms);
+	}
+}
+void VoiceChannel::RemoveRMS(webrtc::AudioRMSInterface* rms) {
+	ASSERT(media_engine_ != NULL);
+	if (media_engine_) {
+		media_engine_->RemoveRMS(rms);
+	}
+}
+webrtc::AudioRMSInterface* VoiceChannel::GetRMS()const {
+	if (media_engine_) {
+		return media_engine_->GetRMS();
+	}
+	return NULL;
+}
+
 void VoiceChannel::OnMessage(rtc::Message *pmsg) {
   switch (pmsg->message_id) {
     case MSG_EARLYMEDIATIMEOUT:
diff --git a/talk/session/media/channel.h b/talk/session/media/channel.h
index 5f51c07..6f1d36e 100644
--- a/talk/session/media/channel.h
+++ b/talk/session/media/channel.h
@@ -49,6 +49,10 @@
 #include "webrtc/base/sigslot.h"
 #include "webrtc/base/window.h"
 
+namespace webrtc {
+	class AudioRMSInterface;
+}
+
 namespace cricket {
 
 struct CryptoParams;
@@ -402,6 +406,11 @@ class VoiceChannel : public BaseChannel {
   // Configuration and setting.
   bool SetChannelOptions(const AudioOptions& options);
 
+  // TODO(dmi): Added by Doubango Telecom
+  virtual void AddRMS(webrtc::AudioRMSInterface* rms);
+  virtual void RemoveRMS(webrtc::AudioRMSInterface* rms);
+  virtual webrtc::AudioRMSInterface* GetRMS()const;
+
  private:
   // overrides from BaseChannel
   virtual void OnChannelRead(TransportChannel* channel,
diff --git a/talk/session/media/channelmanager_unittest.cc b/talk/session/media/channelmanager_unittest.cc
index 1ffdaf2..e0fb43d 100644
--- a/talk/session/media/channelmanager_unittest.cc
+++ b/talk/session/media/channelmanager_unittest.cc
@@ -1,3 +1,4 @@
+
 /*
  * libjingle
  * Copyright 2008 Google Inc.
diff --git a/webrtc/base/win32window.cc b/webrtc/base/win32window.cc
index 4d41014..715f181 100644
--- a/webrtc/base/win32window.cc
+++ b/webrtc/base/win32window.cc
@@ -53,13 +53,21 @@ bool Win32Window::Create(HWND parent, const wchar_t* title, DWORD style,
     wcex.lpfnWndProc = &Win32Window::WndProc;
     wcex.lpszClassName = kWindowBaseClassName;
     window_class_ = ::RegisterClassEx(&wcex);
-    if (!window_class_) {
-      LOG_GLE(LS_ERROR) << "RegisterClassEx failed";
-      return false;
+	if (!window_class_) {
+		if (GetLastError() == ERROR_CLASS_ALREADY_EXISTS) {
+			LOG_GLE(LS_WARNING) << "RegisterClassEx failed: ERROR_CLASS_ALREADY_EXISTS";
+		}
+		else {
+			LOG_GLE(LS_ERROR) << "RegisterClassEx failed";
+			return false;
+		}
     }
   }
   wnd_ = ::CreateWindowEx(exstyle, kWindowBaseClassName, title, style,
                           x, y, cx, cy, parent, NULL, instance_, this);
+  if (wnd_ && !window_class_) {
+	  window_class_ = GetClassWord(wnd_, GCW_ATOM);
+  }
   return (NULL != wnd_);
 }
 
diff --git a/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
index e1a0f69..808c8d1 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
@@ -7,7 +7,7 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-
+#if 0 // TODO(dmi)
 #include "testing/gmock/include/gmock/gmock.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h"
@@ -299,3 +299,4 @@ TEST(NetEqNetworkStatsTest, NoiseExpansionTest) {
 
 
 
+#endif
\ No newline at end of file
diff --git a/webrtc/modules/audio_processing/rms_level.cc b/webrtc/modules/audio_processing/rms_level.cc
index 14136bf..41514fb 100644
--- a/webrtc/modules/audio_processing/rms_level.cc
+++ b/webrtc/modules/audio_processing/rms_level.cc
@@ -39,7 +39,7 @@ void RMSLevel::ProcessMuted(int length) {
   sample_count_ += length;
 }
 
-int RMSLevel::RMS() {
+int RMSLevel::RMS(bool reset /*= true*/) {
   if (sample_count_ == 0 || sum_square_ == 0) {
     Reset();
     return kMinLevel;
@@ -54,7 +54,9 @@ int RMSLevel::RMS() {
     rms = -kMinLevel;
 
   rms = -rms;
-  Reset();
+  if (reset) {
+	  Reset();
+  }
   return static_cast<int>(rms + 0.5);
 }
 
diff --git a/webrtc/modules/audio_processing/rms_level.h b/webrtc/modules/audio_processing/rms_level.h
index 055d271..410a9cf 100644
--- a/webrtc/modules/audio_processing/rms_level.h
+++ b/webrtc/modules/audio_processing/rms_level.h
@@ -44,7 +44,7 @@ class RMSLevel {
   // Computes the RMS level over all data passed to Process() since the last
   // call to RMS(). The returned value is positive but should be interpreted as
   // negative as per the RFC. It is constrained to [0, 127].
-  int RMS();
+  int RMS(bool reset = true);
 
  private:
   float sum_square_;
diff --git a/webrtc/modules/interface/module_common_types.h b/webrtc/modules/interface/module_common_types.h
index 02ce03f..caa7cfd 100644
--- a/webrtc/modules/interface/module_common_types.h
+++ b/webrtc/modules/interface/module_common_types.h
@@ -17,6 +17,7 @@
 #include <algorithm>
 #include <limits>
 
+#include "webrtc/base/refcount.h" // TODO(dmi)
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/common_types.h"
 #include "webrtc/common_video/rotation.h"
@@ -764,6 +765,14 @@ inline uint32_t LatestTimestamp(uint32_t timestamp1, uint32_t timestamp2) {
   return IsNewerTimestamp(timestamp1, timestamp2) ? timestamp1 : timestamp2;
 }
 
+// TODO(dmi): added by Doubango Telcom to implement volume level indicator
+class AudioRMSInterface : public rtc::RefCountInterface{
+public:
+	virtual void RMS(int rms) = 0;
+protected:
+	virtual ~AudioRMSInterface() {}
+};
+
 }  // namespace webrtc
 
 #endif  // MODULE_COMMON_TYPES_H
diff --git a/webrtc/modules/video_capture/windows/video_capture_ds.cc b/webrtc/modules/video_capture/windows/video_capture_ds.cc
index 2d66055..0bbb7dc 100644
--- a/webrtc/modules/video_capture/windows/video_capture_ds.cc
+++ b/webrtc/modules/video_capture/windows/video_capture_ds.cc
@@ -178,7 +178,8 @@ int32_t VideoCaptureDS::StopCapture()
 {
     CriticalSectionScoped cs(&_apiCs);
 
-    HRESULT hr = _mediaControl->Pause();
+	// TODO(dmi) https://code.google.com/p/webrtc/issues/detail?id=1649
+	HRESULT hr = _mediaControl->Stop()/*_mediaControl->Pause()*/;
     if (FAILED(hr))
     {
         WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index 8992425..275d72e 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -161,7 +161,7 @@ Channel::SendData(FrameType frameType,
         // Store current audio level in the RTP/RTCP module.
         // The level will be used in combination with voice-activity state
         // (frameType) to add an RTP header extension
-        _rtpRtcpModule->SetAudioLevel(rms_level_.RMS());
+		_rtpRtcpModule->SetAudioLevel(rms_level_.RMS());
     }
 
     // Push data from ACM to RTP/RTCP-module to deliver audio frame for
@@ -1087,6 +1087,12 @@ Channel::UpdateLocalTimeStamp()
     return 0;
 }
 
+// TODO(dmi): added by Doubango Telecom
+int 
+Channel::RMS() {
+	return rms_level_.RMS(false/*Do not reset, let it done by the RTCP module*/);
+}
+
 int32_t
 Channel::StartPlayout()
 {
@@ -3397,7 +3403,8 @@ Channel::PrepareEncodeAndSend(int mixingFrequency)
 
     InsertInbandDtmfTone();
 
-    if (_includeAudioLevelIndication) {
+	// TODO(dmi): compute rms even if audio level indication not activated
+    /*if (_includeAudioLevelIndication)*/ {
       int length = _audioFrame.samples_per_channel_ * _audioFrame.num_channels_;
       if (is_muted) {
         rms_level_.ProcessMuted(length);
diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h
index 398087e..e7a7da9 100644
--- a/webrtc/voice_engine/channel.h
+++ b/webrtc/voice_engine/channel.h
@@ -185,6 +185,9 @@ public:
 
     // API methods
 
+	// TODO(dmi): added by Doubango Telecom
+	int RMS();
+
     // VoEBase
     int32_t StartPlayout();
     int32_t StopPlayout();
diff --git a/webrtc/voice_engine/include/voe_external_media.h b/webrtc/voice_engine/include/voe_external_media.h
index 6185540..f52ebf7 100644
--- a/webrtc/voice_engine/include/voe_external_media.h
+++ b/webrtc/voice_engine/include/voe_external_media.h
@@ -16,6 +16,7 @@ namespace webrtc {
 
 class VoiceEngine;
 class AudioFrame;
+class AudioRMSInterface;
 
 class WEBRTC_DLLEXPORT VoEMediaProcess {
  public:
@@ -75,6 +76,10 @@ class WEBRTC_DLLEXPORT VoEExternalMedia {
   // Sets the state of external mixing. Cannot be changed during playback.
   virtual int SetExternalMixing(int channel, bool enable) = 0;
 
+  virtual void AddRMS(webrtc::AudioRMSInterface* rms) {  };
+  virtual void RemoveRMS(webrtc::AudioRMSInterface* rms) {  };
+  virtual webrtc::AudioRMSInterface* GetRMS()const { return NULL; };
+
  protected:
   VoEExternalMedia() {}
   virtual ~VoEExternalMedia() {}
diff --git a/webrtc/voice_engine/shared_data.cc b/webrtc/voice_engine/shared_data.cc
index ad00e03..922e8fb 100644
--- a/webrtc/voice_engine/shared_data.cc
+++ b/webrtc/voice_engine/shared_data.cc
@@ -30,8 +30,10 @@ SharedData::SharedData(const Config& config) :
     _engineStatistics(_gInstanceCounter),
     _audioDevicePtr(NULL),
     _moduleProcessThreadPtr(ProcessThread::Create()),
+	_rms(NULL),
     _externalRecording(false),
     _externalPlayout(false)
+	
 {
     Trace::CreateTrace();
     if (OutputMixer::Create(_outputMixerPtr, _gInstanceCounter) == 0)
diff --git a/webrtc/voice_engine/shared_data.h b/webrtc/voice_engine/shared_data.h
index 3ab1d45..f080fef 100644
--- a/webrtc/voice_engine/shared_data.h
+++ b/webrtc/voice_engine/shared_data.h
@@ -12,6 +12,7 @@
 #define WEBRTC_VOICE_ENGINE_SHARED_DATA_H
 
 #include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/utility/interface/process_thread.h"
@@ -24,6 +25,7 @@ class ProcessThread;
 namespace webrtc {
 class Config;
 class CriticalSectionWrapper;
+class AudioRMSInterface;
 
 namespace voe {
 
@@ -56,6 +58,11 @@ public:
       _audioDeviceLayer = layer;
     }
 
+	// TODO(dmi): Added by Doubango Telecom
+	virtual void AddRMS(webrtc::AudioRMSInterface* rms) { _rms = rms; }
+	virtual void RemoveRMS(webrtc::AudioRMSInterface* rms) { _rms = NULL; }
+	virtual webrtc::AudioRMSInterface* GetRMS()const { return _rms; }
+
     int NumOfSendingChannels();
     int NumOfPlayingChannels();
 
@@ -75,6 +82,7 @@ protected:
     TransmitMixer* _transmitMixerPtr;
     rtc::scoped_ptr<AudioProcessing> audioproc_;
     rtc::scoped_ptr<ProcessThread> _moduleProcessThreadPtr;
+	rtc::scoped_refptr<AudioRMSInterface> _rms;
 
     bool _externalRecording;
     bool _externalPlayout;
diff --git a/webrtc/voice_engine/transmit_mixer.cc b/webrtc/voice_engine/transmit_mixer.cc
index edd77b8..2f0bd9b 100644
--- a/webrtc/voice_engine/transmit_mixer.cc
+++ b/webrtc/voice_engine/transmit_mixer.cc
@@ -202,7 +202,8 @@ TransmitMixer::TransmitMixer(uint32_t instanceId) :
     _mute(false),
     _remainingMuteMicTimeMs(0),
     stereo_codec_(false),
-    swap_stereo_channels_(false)
+    swap_stereo_channels_(false),
+	rms_(0)
 {
     WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, -1),
                  "TransmitMixer::TransmitMixer() - ctor");
@@ -409,7 +410,7 @@ TransmitMixer::DemuxAndMix()
 {
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
                  "TransmitMixer::DemuxAndMix()");
-
+	rms_ = 0;
     for (ChannelManager::Iterator it(_channelManagerPtr); it.IsValid();
          it.Increment())
     {
@@ -419,6 +420,7 @@ TransmitMixer::DemuxAndMix()
             // Demultiplex makes a copy of its input.
             channelPtr->Demultiplex(_audioFrame);
             channelPtr->PrepareEncodeAndSend(_audioFrame.sample_rate_hz_);
+			rms_ += channelPtr->RMS();
         }
     }
     return 0;
diff --git a/webrtc/voice_engine/transmit_mixer.h b/webrtc/voice_engine/transmit_mixer.h
index 919de13..7af83e7 100644
--- a/webrtc/voice_engine/transmit_mixer.h
+++ b/webrtc/voice_engine/transmit_mixer.h
@@ -165,6 +165,8 @@ public:
   void EnableStereoChannelSwapping(bool enable);
   bool IsStereoChannelSwappingEnabled();
 
+  int RMS()const { return rms_; } // TODO(dmi): added by Doubango Telecom
+
 private:
     TransmitMixer(uint32_t instanceId);
 
@@ -230,6 +232,7 @@ private:
     bool stereo_codec_;
     bool swap_stereo_channels_;
     rtc::scoped_ptr<int16_t[]> mono_buffer_;
+	int rms_;
 };
 
 }  // namespace voe
diff --git a/webrtc/voice_engine/voe_base_impl.cc b/webrtc/voice_engine/voe_base_impl.cc
index 7bfaddc..df4f999 100644
--- a/webrtc/voice_engine/voe_base_impl.cc
+++ b/webrtc/voice_engine/voe_base_impl.cc
@@ -799,6 +799,13 @@ int VoEBaseImpl::ProcessRecordedDataWithAPM(
                                              number_of_voe_channels);
   }
 
+  // TODO(dmi): Doubango RMS info
+  rtc::scoped_refptr<webrtc::AudioRMSInterface> rms = shared_->GetRMS();
+  if (rms) {
+	  // transmit_mixer RMS value computed in DemuxAndMix()
+	  rms->RMS(shared_->transmit_mixer()->RMS());
+  }
+
   // Scale from VoE to ADM level range.
   uint32_t new_voe_mic_level = shared_->transmit_mixer()->CaptureLevel();
   if (new_voe_mic_level != voe_mic_level) {
diff --git a/webrtc/voice_engine/voice_engine_impl.cc b/webrtc/voice_engine/voice_engine_impl.cc
index f7f363d..33c2077 100644
--- a/webrtc/voice_engine/voice_engine_impl.cc
+++ b/webrtc/voice_engine/voice_engine_impl.cc
@@ -57,6 +57,17 @@ int VoiceEngineImpl::AddRef() {
   return ++_ref_count;
 }
 
+// TODO(dmi): Added by Doubango Telecom
+void VoiceEngineImpl::AddRMS(webrtc::AudioRMSInterface* rms) {
+	SharedData::AddRMS(rms); // call base class function
+}
+void VoiceEngineImpl::RemoveRMS(webrtc::AudioRMSInterface* rms) {
+	SharedData::RemoveRMS(rms); // call base class function
+}
+webrtc::AudioRMSInterface* VoiceEngineImpl::GetRMS()const {
+	return SharedData::GetRMS(); // call base class function
+}
+
 // This implements the Release() method for all the inherited interfaces.
 int VoiceEngineImpl::Release() {
   int new_ref = --_ref_count;
diff --git a/webrtc/voice_engine/voice_engine_impl.h b/webrtc/voice_engine/voice_engine_impl.h
index 07f29c3..9299438 100644
--- a/webrtc/voice_engine/voice_engine_impl.h
+++ b/webrtc/voice_engine/voice_engine_impl.h
@@ -48,6 +48,7 @@
 #endif
 
 namespace webrtc {
+	class AudioRMSInterface;
 
 class VoiceEngineImpl : public voe::SharedData,  // Must be the first base class
                         public VoiceEngine,
@@ -128,6 +129,11 @@ class VoiceEngineImpl : public voe::SharedData,  // Must be the first base class
   // This implements the Release() method for all the inherited interfaces.
   int Release() override;
 
+  // TODO(dmi): Added by Doubango Telecom
+  virtual void AddRMS(webrtc::AudioRMSInterface* rms);
+  virtual void RemoveRMS(webrtc::AudioRMSInterface* rms);
+  virtual webrtc::AudioRMSInterface* GetRMS()const;
+
  private:
   Atomic32 _ref_count;
   rtc::scoped_ptr<const Config> own_config_;