Backout df75a87cce60 & 19e164f7d88d (bug 818670) for build bustage on a CLOSED TREE

content/media/webrtc/MediaEngine.h

@@ -89,11 +89,6 @@ public:
   /* Stop the device and release the corresponding MediaStream */
   virtual nsresult Stop(SourceMediaStream *aSource, TrackID aID) = 0;
 
-  /* Change device configuration.  */
-  virtual nsresult Config(bool aEchoOn, uint32_t aEcho,
-                          bool aAgcOn, uint32_t aAGC,
-                          bool aNoiseOn, uint32_t aNoise) = 0;
-
   /* Return false if device is currently allocated or started */
   bool IsAvailable() {
     if (mState == kAllocated || mState == kStarted) {

content/media/webrtc/MediaEngineDefault.h

@@ -44,9 +44,6 @@ public:
   virtual nsresult Start(SourceMediaStream*, TrackID);
   virtual nsresult Stop(SourceMediaStream*, TrackID);
   virtual nsresult Snapshot(uint32_t aDuration, nsIDOMFile** aFile);
-  virtual nsresult Config(bool aEchoOn, uint32_t aEcho,
-                          bool aAgcOn, uint32_t aAGC,
-                          bool aNoiseOn, uint32_t aNoise) { return NS_OK; };
   virtual void NotifyPull(MediaStreamGraph* aGraph, StreamTime aDesiredTime);
   virtual void NotifyPull(MediaStreamGraph* aGraph,
                           SourceMediaStream *aSource,
@@ -89,9 +86,6 @@ public:
   virtual nsresult Start(SourceMediaStream*, TrackID);
   virtual nsresult Stop(SourceMediaStream*, TrackID);
   virtual nsresult Snapshot(uint32_t aDuration, nsIDOMFile** aFile);
-  virtual nsresult Config(bool aEchoOn, uint32_t aEcho,
-                          bool aAgcOn, uint32_t aAGC,
-                          bool aNoiseOn, uint32_t aNoise) { return NS_OK; };
   virtual void NotifyPull(MediaStreamGraph* aGraph, StreamTime aDesiredTime);
   virtual void NotifyPull(MediaStreamGraph* aGraph,
                           SourceMediaStream *aSource,

content/media/webrtc/MediaEngineWebRTC.h

@@ -35,7 +35,6 @@
 #include "voice_engine/include/voe_audio_processing.h"
 #include "voice_engine/include/voe_volume_control.h"
 #include "voice_engine/include/voe_external_media.h"
-#include "voice_engine/include/voe_audio_processing.h"
 
 // Video Engine
 #include "video_engine/include/vie_base.h"
@@ -91,9 +90,6 @@ public:
   virtual nsresult Start(SourceMediaStream*, TrackID);
   virtual nsresult Stop(SourceMediaStream*, TrackID);
   virtual nsresult Snapshot(uint32_t aDuration, nsIDOMFile** aFile);
-  virtual nsresult Config(bool aEchoOn, uint32_t aEcho,
-                          bool aAgcOn, uint32_t aAGC,
-                          bool aNoiseOn, uint32_t aNoise) { return NS_OK; };
   virtual void NotifyPull(MediaStreamGraph* aGraph,
                           SourceMediaStream *aSource,
                           TrackID aId,
@@ -181,10 +177,6 @@ public:
     , mCapIndex(aIndex)
     , mChannel(-1)
     , mInitDone(false)
-    , mEchoOn(false), mAgcOn(false), mNoiseOn(false)
-    , mEchoCancel(webrtc::kEcDefault)
-    , mAGC(webrtc::kAgcDefault)
-    , mNoiseSuppress(webrtc::kNsDefault)
     , mNullTransport(nullptr) {
     MOZ_ASSERT(aVoiceEnginePtr);
     mState = kReleased;
@@ -202,10 +194,6 @@ public:
   virtual nsresult Start(SourceMediaStream*, TrackID);
   virtual nsresult Stop(SourceMediaStream*, TrackID);
   virtual nsresult Snapshot(uint32_t aDuration, nsIDOMFile** aFile);
-  virtual nsresult Config(bool aEchoOn, uint32_t aEcho,
-                          bool aAgcOn, uint32_t aAGC,
-                          bool aNoiseOn, uint32_t aNoise);
-
   virtual void NotifyPull(MediaStreamGraph* aGraph,
                           SourceMediaStream *aSource,
                           TrackID aId,
@@ -230,7 +218,6 @@ private:
   webrtc::VoEBase* mVoEBase;
   webrtc::VoEExternalMedia* mVoERender;
   webrtc::VoENetwork*  mVoENetwork;
-  webrtc::VoEAudioProcessing *mVoEProcessing;
 
   // mMonitor protects mSources[] access/changes, and transitions of mState
   // from kStarted to kStopped (which are combined with EndTrack()).
@@ -246,11 +233,6 @@ private:
   nsString mDeviceName;
   nsString mDeviceUUID;
 
-  bool mEchoOn, mAgcOn, mNoiseOn;
-  webrtc::EcModes  mEchoCancel;
-  webrtc::AgcModes mAGC;
-  webrtc::NsModes  mNoiseSuppress;
-
   NullTransport *mNullTransport;
 };
 

content/media/webrtc/MediaEngineWebRTCAudio.cpp

@@ -46,60 +46,6 @@ MediaEngineWebRTCAudioSource::GetUUID(nsAString& aUUID)
   return;
 }
 
-nsresult
-MediaEngineWebRTCAudioSource::Config(bool aEchoOn, uint32_t aEcho,
-                                     bool aAgcOn, uint32_t aAGC,
-                                     bool aNoiseOn, uint32_t aNoise)
-{
-  LOG(("Audio config: aec: %d, agc: %d, noise: %d",
-       aEchoOn ? aEcho : -1,
-       aAgcOn ? aAGC : -1,
-       aNoiseOn ? aNoise : -1));
-
-  bool update_agc = (mAgcOn == aAgcOn);
-  bool update_noise = (mNoiseOn == aNoiseOn);
-  mAgcOn = aAgcOn;
-  mNoiseOn = aNoiseOn;
-
-  if ((webrtc::AgcModes) aAGC != webrtc::kAgcUnchanged) {
-    if (mAGC != (webrtc::AgcModes) aAGC) {
-      update_agc = true;
-      mAGC = (webrtc::AgcModes) aAGC;
-    }
-  }
-  if ((webrtc::NsModes) aNoise != webrtc::kNsUnchanged) {
-    if (mNoiseSuppress != (webrtc::NsModes) aNoise) {
-      update_noise = true;
-      mNoiseSuppress = (webrtc::NsModes) aNoise;
-    }
-  }
-
-  if (mInitDone) {
-    int error;
-#if 0
-    // Until we can support feeding our full output audio from the browser
-    // through the MediaStream, this won't work.  Or we need to move AEC to
-    // below audio input and output, perhaps invoked from here.
-    mEchoOn = aEchoOn;
-    if ((webrtc::EcModes) aEcho != webrtc::kEcUnchanged)
-      mEchoCancel = (webrtc::EcModes) aEcho;
-    mVoEProcessing->SetEcStatus(mEchoOn, aEcho);
-#else
-    (void) aEcho; (void) aEchoOn; // suppress warnings
-#endif
-
-    if (update_agc &&
-      0 != (error = mVoEProcessing->SetAgcStatus(mAgcOn, (webrtc::AgcModes) aAGC))) {
-      LOG(("%s Error setting AGC Status: %d ",__FUNCTION__, error));
-    }
-    if (update_noise &&
-      0 != (error = mVoEProcessing->SetNsStatus(mNoiseOn, (webrtc::NsModes) aNoise))) {
-      LOG(("%s Error setting NoiseSuppression Status: %d ",__FUNCTION__, error));
-    }
-  }
-  return NS_OK;
-}
-
 nsresult
 MediaEngineWebRTCAudioSource::Allocate()
 {
@@ -160,11 +106,6 @@ MediaEngineWebRTCAudioSource::Start(SourceMediaStream* aStream, TrackID aID)
   }
   mState = kStarted;
 
-  // Configure audio processing in webrtc code
-  Config(mEchoOn, webrtc::kEcUnchanged,
-         mAgcOn, webrtc::kAgcUnchanged,
-         mNoiseOn, webrtc::kNsUnchanged);
-
   if (mVoEBase->StartReceive(mChannel)) {
     return NS_ERROR_FAILURE;
   }
@@ -251,11 +192,6 @@ MediaEngineWebRTCAudioSource::Init()
     return;
   }
 
-  mVoEProcessing = webrtc::VoEAudioProcessing::GetInterface(mVoiceEngine);
-  if (!mVoEProcessing) {
-    return;
-  }
-
   mChannel = mVoEBase->CreateChannel();
   if (mChannel < 0) {
     return;

dom/media/MediaManager.cpp

@@ -11,8 +11,6 @@
 #include "nsIScriptGlobalObject.h"
 #include "nsIPopupWindowManager.h"
 #include "nsISupportsArray.h"
-#include "nsIPrefService.h"
-#include "nsIPrefBranch.h"
 
 // For PR_snprintf
 #include "prprf.h"
@@ -384,30 +382,6 @@ public:
                                  mAudioSource, mVideoSource, false));
     mediaThread->Dispatch(runnable, NS_DISPATCH_NORMAL);
 
-    nsresult rv;
-    nsCOMPtr<nsIPrefService> prefs = do_GetService("@mozilla.org/preferences-service;1", &rv);
-    if (NS_SUCCEEDED(rv)) {
-      nsCOMPtr<nsIPrefBranch> branch = do_QueryInterface(prefs);
-
-      if (branch) {
-        int32_t aec = (int32_t) webrtc::kEcUnchanged;
-        int32_t agc = (int32_t) webrtc::kAgcUnchanged;
-        int32_t noise = (int32_t) webrtc::kNsUnchanged;
-        bool aec_on = false, agc_on = false, noise_on = false;
-
-        branch->GetBoolPref("media.peerconnection.aec_enabled", &aec_on);
-        branch->GetIntPref("media.peerconnection.aec", &aec);
-        branch->GetBoolPref("media.peerconnection.agc_enabled", &agc_on);
-        branch->GetIntPref("media.peerconnection.agc", &agc);
-        branch->GetBoolPref("media.peerconnection.noise_enabled", &noise_on);
-        branch->GetIntPref("media.peerconnection.noise", &noise);
-
-        mListener->AudioConfig(aec_on, (uint32_t) aec,
-                               agc_on, (uint32_t) agc,
-                               noise_on, (uint32_t) noise);
-      }
-    }
-
     // We're in the main thread, so no worries here either.
     nsCOMPtr<nsIDOMGetUserMediaSuccessCallback> success(mSuccess);
     nsCOMPtr<nsIDOMGetUserMediaErrorCallback> error(mError);

dom/media/MediaManager.h

@@ -21,8 +21,6 @@
 #include "mozilla/StaticPtr.h"
 #include "prlog.h"
 
-#include "mtransport/runnable_utils.h"
-
 namespace mozilla {
 
 #ifdef PR_LOGGING
@@ -130,20 +128,6 @@ public:
   // Can be invoked from EITHER MainThread or MSG thread
   void Invalidate();
 
-  void
-  AudioConfig(bool aEchoOn, uint32_t aEcho,
-              bool aAgcOn, uint32_t aAGC,
-              bool aNoiseOn, uint32_t aNoise)
-  {
-    if (mAudioSource) {
-      RUN_ON_THREAD(mMediaThread,
-                    WrapRunnable(nsRefPtr<MediaEngineSource>(mAudioSource), // threadsafe
-                                 &MediaEngineSource::Config,
-                                 aEchoOn, aEcho, aAgcOn, aAGC, aNoiseOn, aNoise),
-                    NS_DISPATCH_NORMAL);
-    }
-  }
-
   void
   Remove()
   {

media/mtransport/build/Makefile.in

@@ -37,7 +37,6 @@ EXPORTS_mtransport = \
   ../transportlayerprsock.h \
   ../m_cpp_utils.h \
   ../runnable_utils.h \
-  ../runnable_utils_generated.h \
   ../sigslot.h \
   $(NULL)
 

media/webrtc/signaling/src/media-conduit/AudioConduit.cpp

@@ -3,13 +3,6 @@
  * You can obtain one at http://mozilla.org/MPL/2.0/. */
 
 #include "AudioConduit.h"
-#include "nsCOMPtr.h"
-#include "mozilla/Services.h"
-#include "nsServiceManagerUtils.h"
-#include "nsIPrefService.h"
-#include "nsIPrefBranch.h"
-#include "nsThreadUtils.h"
-
 #include "CSFLog.h"
 #include "voice_engine/include/voe_errors.h"
 
@@ -24,13 +17,11 @@ const unsigned int WebrtcAudioConduit::CODEC_PLNAME_SIZE = 32;
 /**
  * Factory Method for AudioConduit
  */
-mozilla::RefPtr<AudioSessionConduit> AudioSessionConduit::Create(AudioSessionConduit *aOther)
+mozilla::RefPtr<AudioSessionConduit> AudioSessionConduit::Create()
 {
   CSFLogDebug(logTag,  "%s ", __FUNCTION__);
-  NS_ASSERTION(NS_IsMainThread(), "Only call on main thread");
-
   WebrtcAudioConduit* obj = new WebrtcAudioConduit();
-  if(obj->Init(static_cast<WebrtcAudioConduit*>(aOther)) != kMediaConduitNoError)
+  if(obj->Init() != kMediaConduitNoError)
   {
     CSFLogError(logTag,  "%s AudioConduit Init Failed ", __FUNCTION__);
     delete obj;
@@ -45,8 +36,6 @@ mozilla::RefPtr<AudioSessionConduit> AudioSessionConduit::Create(AudioSessionCon
  */
 WebrtcAudioConduit::~WebrtcAudioConduit()
 {
-  NS_ASSERTION(NS_IsMainThread(), "Only call on main thread");
-
   CSFLogDebug(logTag,  "%s ", __FUNCTION__);
   for(std::vector<AudioCodecConfig*>::size_type i=0;i < mRecvCodecList.size();i++)
   {
@@ -55,27 +44,17 @@ WebrtcAudioConduit::~WebrtcAudioConduit()
 
   delete mCurSendCodecConfig;
 
-  // The first one of a pair to be deleted shuts down media for both
   if(mPtrVoEXmedia)
   {
-    if (!mShutDown) {
+    mPtrVoEXmedia->SetExternalRecordingStatus(false);
-      mPtrVoEXmedia->SetExternalRecordingStatus(false);
+    mPtrVoEXmedia->SetExternalPlayoutStatus(false);
-      mPtrVoEXmedia->SetExternalPlayoutStatus(false);
-    }
     mPtrVoEXmedia->Release();
   }
 
-  if(mPtrVoEProcessing)
-  {
-    mPtrVoEProcessing->Release();
-  }
-
   //Deal with the transport
   if(mPtrVoENetwork)
   {
-    if (!mShutDown) {
+    mPtrVoENetwork->DeRegisterExternalTransport(mChannel);
-      mPtrVoENetwork->DeRegisterExternalTransport(mChannel);
-    }
     mPtrVoENetwork->Release();
   }
 
@@ -86,69 +65,47 @@ WebrtcAudioConduit::~WebrtcAudioConduit()
 
   if(mPtrVoEBase)
   {
-    if (!mShutDown) {
+    mPtrVoEBase->StopPlayout(mChannel);
-      mPtrVoEBase->StopPlayout(mChannel);
+    mPtrVoEBase->StopSend(mChannel);
-      mPtrVoEBase->StopSend(mChannel);
+    mPtrVoEBase->StopReceive(mChannel);
-      mPtrVoEBase->StopReceive(mChannel);
+    mPtrVoEBase->DeleteChannel(mChannel);
-      mPtrVoEBase->DeleteChannel(mChannel);
+    mPtrVoEBase->Terminate();
-      mPtrVoEBase->Terminate();
-    }
     mPtrVoEBase->Release();
   }
 
-  if (mOtherDirection)
+  if(mVoiceEngine)
   {
-    // mOtherDirection owns these now!
+    webrtc::VoiceEngine::Delete(mVoiceEngine);
-    mOtherDirection->mOtherDirection = NULL;
-    // let other side we terminated the channel
-    mOtherDirection->mShutDown = true;
-    mVoiceEngine = nullptr;
-  } else {
-    // only one opener can call Delete.  Have it be the last to close.
-    if(mVoiceEngine)
-    {
-      webrtc::VoiceEngine::Delete(mVoiceEngine);
-    }
   }
 }
 
 /*
  * WebRTCAudioConduit Implementation
  */
-MediaConduitErrorCode WebrtcAudioConduit::Init(WebrtcAudioConduit *other)
+MediaConduitErrorCode WebrtcAudioConduit::Init()
 {
-  CSFLogDebug(logTag,  "%s this=%p other=%p", __FUNCTION__, this, other);
+  CSFLogDebug(logTag,  "%s ", __FUNCTION__);
 
-  if (other) {
+  //Per WebRTC APIs below function calls return NULL on failure
-    MOZ_ASSERT(!other->mOtherDirection);
+  if(!(mVoiceEngine = webrtc::VoiceEngine::Create()))
-    other->mOtherDirection = this;
+  {
-    mOtherDirection = other;
+    CSFLogError(logTag, "%s Unable to create voice engine", __FUNCTION__);
+    return kMediaConduitSessionNotInited;
+  }
 
-    // only one can call ::Create()/GetVoiceEngine()
+  PRLogModuleInfo *logs = GetWebRTCLogInfo();
-    MOZ_ASSERT(other->mVoiceEngine);
+  if (!gWebrtcTraceLoggingOn && logs && logs->level > 0) {
-    mVoiceEngine = other->mVoiceEngine;
+    // no need to a critical section or lock here
-  } else {
+    gWebrtcTraceLoggingOn = 1;
-    //Per WebRTC APIs below function calls return NULL on failure
+
-    if(!(mVoiceEngine = webrtc::VoiceEngine::Create()))
+    const char *file = PR_GetEnv("WEBRTC_TRACE_FILE");
-    {
+    if (!file) {
-      CSFLogError(logTag, "%s Unable to create voice engine", __FUNCTION__);
+      file = "WebRTC.log";
-      return kMediaConduitSessionNotInited;
-    }
-
-    PRLogModuleInfo *logs = GetWebRTCLogInfo();
-    if (!gWebrtcTraceLoggingOn && logs && logs->level > 0) {
-      // no need to a critical section or lock here
-      gWebrtcTraceLoggingOn = 1;
-
-      const char *file = PR_GetEnv("WEBRTC_TRACE_FILE");
-      if (!file) {
-        file = "WebRTC.log";
-      }
-      CSFLogDebug(logTag,  "%s Logging webrtc to %s level %d", __FUNCTION__,
-                  file, logs->level);
-      mVoiceEngine->SetTraceFilter(logs->level);
-      mVoiceEngine->SetTraceFile(file);
     }
+    CSFLogDebug(logTag,  "%s Logging webrtc to %s level %d", __FUNCTION__,
+                file, logs->level);
+    mVoiceEngine->SetTraceFilter(logs->level);
+    mVoiceEngine->SetTraceFile(file);
   }
 
   if(!(mPtrVoEBase = VoEBase::GetInterface(mVoiceEngine)))
@@ -169,60 +126,51 @@ MediaConduitErrorCode WebrtcAudioConduit::Init(WebrtcAudioConduit *other)
     return kMediaConduitSessionNotInited;
   }
 
-  if(!(mPtrVoEProcessing = VoEAudioProcessing::GetInterface(mVoiceEngine)))
-  {
-    CSFLogError(logTag, "%s Unable to initialize VoEProcessing", __FUNCTION__);
-    return kMediaConduitSessionNotInited;
-  }
-
   if(!(mPtrVoEXmedia = VoEExternalMedia::GetInterface(mVoiceEngine)))
   {
     CSFLogError(logTag, "%s Unable to initialize VoEExternalMedia", __FUNCTION__);
     return kMediaConduitSessionNotInited;
   }
 
-  if (other) {
+  // init the engine with our audio device layer
-    mChannel = other->mChannel;
+  if(mPtrVoEBase->Init() == -1)
-  } else {
+  {
-    // init the engine with our audio device layer
+    CSFLogError(logTag, "%s VoiceEngine Base Not Initialized", __FUNCTION__);
-    if(mPtrVoEBase->Init() == -1)
+    return kMediaConduitSessionNotInited;
-    {
-      CSFLogError(logTag, "%s VoiceEngine Base Not Initialized", __FUNCTION__);
-      return kMediaConduitSessionNotInited;
-    }
-
-    if( (mChannel = mPtrVoEBase->CreateChannel()) == -1)
-    {
-      CSFLogError(logTag, "%s VoiceEngine Channel creation failed",__FUNCTION__);
-      return kMediaConduitChannelError;
-    }
-
-    CSFLogDebug(logTag, "%s Channel Created %d ",__FUNCTION__, mChannel);
-
-    if(mPtrVoENetwork->RegisterExternalTransport(mChannel, *this) == -1)
-    {
-      CSFLogError(logTag, "%s VoiceEngine, External Transport Failed",__FUNCTION__);
-      return kMediaConduitTransportRegistrationFail;
-    }
-
-    if(mPtrVoEXmedia->SetExternalRecordingStatus(true) == -1)
-    {
-      CSFLogError(logTag, "%s SetExternalRecordingStatus Failed %d",__FUNCTION__,
-                  mPtrVoEBase->LastError());
-      return kMediaConduitExternalPlayoutError;
-    }
-
-    if(mPtrVoEXmedia->SetExternalPlayoutStatus(true) == -1)
-    {
-      CSFLogError(logTag, "%s SetExternalPlayoutStatus Failed %d ",__FUNCTION__,
-                  mPtrVoEBase->LastError());
-      return kMediaConduitExternalRecordingError;
-    }
-    CSFLogDebug(logTag ,  "%s AudioSessionConduit Initialization Done (%p)",__FUNCTION__, this);
   }
+
+  if( (mChannel = mPtrVoEBase->CreateChannel()) == -1)
+  {
+    CSFLogError(logTag, "%s VoiceEngine Channel creation failed",__FUNCTION__);
+    return kMediaConduitChannelError;
+  }
+
+  CSFLogDebug(logTag, "%s Channel Created %d ",__FUNCTION__, mChannel);
+
+  if(mPtrVoENetwork->RegisterExternalTransport(mChannel, *this) == -1)
+  {
+    CSFLogError(logTag, "%s VoiceEngine, External Transport Failed",__FUNCTION__);
+    return kMediaConduitTransportRegistrationFail;
+  }
+
+  if(mPtrVoEXmedia->SetExternalRecordingStatus(true) == -1)
+  {
+    CSFLogError(logTag, "%s SetExternalRecordingStatus Failed %d",__FUNCTION__,
+                                                      mPtrVoEBase->LastError());
+    return kMediaConduitExternalPlayoutError;
+  }
+
+  if(mPtrVoEXmedia->SetExternalPlayoutStatus(true) == -1)
+  {
+    CSFLogError(logTag, "%s SetExternalPlayoutStatus Failed %d ",__FUNCTION__,
+                                                     mPtrVoEBase->LastError());
+    return kMediaConduitExternalRecordingError;
+  }
+  CSFLogDebug(logTag ,  "%s AudioSessionConduit Initialization Done",__FUNCTION__);
   return kMediaConduitNoError;
 }
 
+
 // AudioSessionConduit Implementation
 MediaConduitErrorCode
 WebrtcAudioConduit::AttachTransport(mozilla::RefPtr<TransportInterface> aTransport)
@@ -287,33 +235,6 @@ WebrtcAudioConduit::ConfigureSendMediaCodec(const AudioCodecConfig* codecConfig)
     return kMediaConduitUnknownError;
   }
 
-  // TEMPORARY - see bug 694814 comment 2
-  nsresult rv;
-  nsCOMPtr<nsIPrefService> prefs = do_GetService("@mozilla.org/preferences-service;1", &rv);
-  if (NS_SUCCEEDED(rv)) {
-    nsCOMPtr<nsIPrefBranch> branch = do_QueryInterface(prefs);
-
-    if (branch) {
-      int32_t aec = 0; // 0 == unchanged
-      bool aec_on = false;
-
-      branch->GetBoolPref("media.peerconnection.aec_enabled", &aec_on);
-      branch->GetIntPref("media.peerconnection.aec", &aec);
-
-      CSFLogDebug(logTag,"Audio config: aec: %d", aec_on ? aec : -1);
-      mEchoOn = aec_on;
-      if (static_cast<webrtc::EcModes>(aec) != webrtc::kEcUnchanged)
-        mEchoCancel = static_cast<webrtc::EcModes>(aec);
-
-      branch->GetIntPref("media.peerconnection.capture_delay", &mCaptureDelay);
-    }
-  }
-
-  if (0 != (error = mPtrVoEProcessing->SetEcStatus(mEchoOn, mEchoCancel))) {
-    CSFLogError(logTag,"%s Error setting EVStatus: %d ",__FUNCTION__, error);
-    return kMediaConduitUnknownError;
-  }
-
   //Let's Send Transport State-machine on the Engine
   if(mPtrVoEBase->StartSend(mChannel) == -1)
   {
@@ -483,7 +404,7 @@ WebrtcAudioConduit::SendAudioFrame(const int16_t audio_data[],
     return kMediaConduitSessionNotInited;
   }
 
-  capture_delay = mCaptureDelay;
+
   //Insert the samples
   if(mPtrVoEXmedia->ExternalRecordingInsertData(audio_data,
                                                 lengthSamples,
@@ -623,53 +544,34 @@ WebrtcAudioConduit::ReceivedRTCPPacket(const void *data, int len)
 
 int WebrtcAudioConduit::SendPacket(int channel, const void* data, int len)
 {
-  CSFLogDebug(logTag,  "%s : channel %d %s",__FUNCTION__,channel,
+  CSFLogDebug(logTag,  "%s : channel %d",__FUNCTION__,channel);
-              (mEngineReceiving && mOtherDirection) ? "(using mOtherDirection)" : "");
 
-  if (mEngineReceiving)
+   if(mTransport && (mTransport->SendRtpPacket(data, len) == NS_OK))
-  {
+   {
-    if (mOtherDirection)
-    {
-      return mOtherDirection->SendPacket(channel, data, len);
-    }
-    CSFLogDebug(logTag,  "%s : Asked to send RTP without an RTP sender",
-                __FUNCTION__, channel);
-    return -1;
-  } else {
-    if(mTransport && (mTransport->SendRtpPacket(data, len) == NS_OK))
-    {
       CSFLogDebug(logTag, "%s Sent RTP Packet ", __FUNCTION__);
       return len;
-    } else {
+   } else {
-      CSFLogError(logTag, "%s RTP Packet Send Failed ", __FUNCTION__);
+     CSFLogError(logTag, "%s RTP Packet Send Failed ", __FUNCTION__);
-      return -1;
+     return -1;
-    }
+   }
-  }
+
 }
 
 int WebrtcAudioConduit::SendRTCPPacket(int channel, const void* data, int len)
 {
   CSFLogDebug(logTag,  "%s : channel %d", __FUNCTION__, channel);
 
-  if (mEngineTransmitting)
+  // can't enable this assertion, because we do.  Suppress it
+  // NS_ASSERTION(mEngineReceiving,"We shouldn't send RTCP on the receiver side");
+  if(mEngineReceiving && mTransport && mTransport->SendRtcpPacket(data, len) == NS_OK)
   {
-    if (mOtherDirection)
+    CSFLogDebug(logTag, "%s Sent RTCP Packet ", __FUNCTION__);
-    {
+    return len;
-      return mOtherDirection->SendRTCPPacket(channel, data, len);
-    }
-    CSFLogDebug(logTag,  "%s : Asked to send RTCP without an RTP receiver",
-                __FUNCTION__, channel);
-    return -1;
   } else {
-    if(mTransport && mTransport->SendRtcpPacket(data, len) == NS_OK)
+    CSFLogError(logTag, "%s RTCP Packet Send Failed ", __FUNCTION__);
-    {
+    return -1;
-      CSFLogDebug(logTag, "%s Sent RTCP Packet ", __FUNCTION__);
-      return len;
-    } else {
-      CSFLogError(logTag, "%s RTCP Packet Send Failed ", __FUNCTION__);
-      return -1;
-    }
   }
+
 }
 
 /**
@@ -846,3 +748,4 @@ WebrtcAudioConduit::DumpCodecDB() const
     }
  }
 }// end namespace
+

media/webrtc/signaling/src/media-conduit/AudioConduit.h

@@ -18,14 +18,12 @@
 #include "voice_engine/include/voe_file.h"
 #include "voice_engine/include/voe_network.h"
 #include "voice_engine/include/voe_external_media.h"
-#include "voice_engine/include/voe_audio_processing.h"
 
 //Some WebRTC types for short notations
  using webrtc::VoEBase;
  using webrtc::VoENetwork;
  using webrtc::VoECodec;
  using webrtc::VoEExternalMedia;
- using webrtc::VoEAudioProcessing;
 
 /** This file hosts several structures identifying different aspects
  * of a RTP Session.
@@ -143,23 +141,18 @@ public:
 
 
   WebrtcAudioConduit():
-                      mOtherDirection(NULL),
-                      mShutDown(false),
                       mVoiceEngine(NULL),
                       mTransport(NULL),
                       mEngineTransmitting(false),
                       mEngineReceiving(false),
                       mChannel(-1),
-                      mCurSendCodecConfig(NULL),
+                      mCurSendCodecConfig(NULL)
-                      mCaptureDelay(150),
-                      mEchoOn(true),
-                      mEchoCancel(webrtc::kEcAec)
   {
   }
 
   virtual ~WebrtcAudioConduit();
 
-  MediaConduitErrorCode Init(WebrtcAudioConduit *other);
+  MediaConduitErrorCode Init();
 
 private:
   WebrtcAudioConduit(const WebrtcAudioConduit& other) MOZ_DELETE;
@@ -192,19 +185,12 @@ private:
   //Utility function to dump recv codec database
   void DumpCodecDB() const;
 
-  WebrtcAudioConduit*  mOtherDirection;
-  // Other side has shut down our channel and related items already
-  bool mShutDown;
-
-  // These are shared by both directions.  They're released by the last
-  // conduit to die
   webrtc::VoiceEngine* mVoiceEngine;
   mozilla::RefPtr<TransportInterface> mTransport;
   webrtc::VoENetwork*  mPtrVoENetwork;
   webrtc::VoEBase*     mPtrVoEBase;
   webrtc::VoECodec*    mPtrVoECodec;
   webrtc::VoEExternalMedia* mPtrVoEXmedia;
-  webrtc::VoEAudioProcessing* mPtrVoEProcessing;
 
   //engine states of our interets
   bool mEngineTransmitting; // If true => VoiceEngine Send-subsystem is up
@@ -214,12 +200,6 @@ private:
   int mChannel;
   RecvCodecList    mRecvCodecList;
   AudioCodecConfig* mCurSendCodecConfig;
-
-  // Current "capture" delay (really output plus input delay)
-  int32_t mCaptureDelay;
-
-  bool mEchoOn;
-  webrtc::EcModes  mEchoCancel;
 };
 
 } // end namespace

media/webrtc/signaling/src/media-conduit/MediaConduitInterface.h

@@ -226,7 +226,7 @@ public:
     * return: Concrete VideoSessionConduitObject or NULL in the case
     *         of failure
     */
-  static mozilla::RefPtr<AudioSessionConduit> Create(AudioSessionConduit *aOther);
+  static mozilla::RefPtr<AudioSessionConduit> Create();
 
   virtual ~AudioSessionConduit() {}
 

media/webrtc/signaling/src/media/VcmSIPCCBinding.cpp

@@ -1303,18 +1303,12 @@ static int vcmRxStartICE_m(cc_mcapid_t mcap_id,
 
   if (CC_IS_AUDIO(mcap_id)) {
     std::vector<mozilla::AudioCodecConfig *> configs;
-
     // Instantiate an appropriate conduit
-    mozilla::RefPtr<mozilla::AudioSessionConduit> tx_conduit =
-      pc.impl()->media()->GetConduit(level, false);
-
     mozilla::RefPtr<mozilla::AudioSessionConduit> conduit =
-                    mozilla::AudioSessionConduit::Create(tx_conduit);
+                    mozilla::AudioSessionConduit::Create();
     if(!conduit)
       return VCM_ERROR;
 
-    pc.impl()->media()->AddConduit(level, true, conduit);
-
     mozilla::AudioCodecConfig *config_raw;
 
     for(int i=0; i <num_payloads ; i++)
@@ -1959,17 +1953,12 @@ static int vcmTxStartICE_m(cc_mcapid_t mcap_id,
     mozilla::ScopedDeletePtr<mozilla::AudioCodecConfig> config(config_raw);
 
     // Instantiate an appropriate conduit
-    mozilla::RefPtr<mozilla::AudioSessionConduit> rx_conduit =
-      pc.impl()->media()->GetConduit(level, true);
-
     mozilla::RefPtr<mozilla::AudioSessionConduit> conduit =
-      mozilla::AudioSessionConduit::Create(rx_conduit);
+      mozilla::AudioSessionConduit::Create();
 
     if (!conduit || conduit->ConfigureSendMediaCodec(config))
       return VCM_ERROR;
 
-    pc.impl()->media()->AddConduit(level, false, conduit);
-
     mozilla::RefPtr<mozilla::MediaPipeline> pipeline =
         new mozilla::MediaPipelineTransmit(
             pc.impl()->GetHandle(),

media/webrtc/signaling/src/peerconnection/PeerConnectionMedia.h

@@ -293,28 +293,9 @@ class PeerConnectionMedia : public sigslot::has_slots<> {
                         mozilla::RefPtr<mozilla::TransportFlow> aFlow) {
     int index_inner = aIndex * 2 + (aRtcp ? 1 : 0);
 
-    MOZ_ASSERT(!mTransportFlows[index_inner]);
     mTransportFlows[index_inner] = aFlow;
   }
 
-  mozilla::RefPtr<mozilla::AudioSessionConduit> GetConduit(int aStreamIndex, bool aReceive) {
-    int index_inner = aStreamIndex * 2 + (aReceive ? 0 : 1);
-
-    if (mAudioConduits.find(index_inner) == mAudioConduits.end())
-      return NULL;
-
-    return mAudioConduits[index_inner];
-  }
-
-  // Add a conduit
-  void AddConduit(int aIndex, bool aReceive,
-                  const mozilla::RefPtr<mozilla::AudioSessionConduit> &aConduit) {
-    int index_inner = aIndex * 2 + (aReceive ? 0 : 1);
-
-    MOZ_ASSERT(!mAudioConduits[index_inner]);
-    mAudioConduits[index_inner] = aConduit;
-  }
-
   // ICE state signals
   sigslot::signal1<mozilla::NrIceCtx *> SignalIceGatheringCompleted;  // Done gathering
   sigslot::signal1<mozilla::NrIceCtx *> SignalIceCompleted;  // Done handshaking
@@ -350,10 +331,6 @@ class PeerConnectionMedia : public sigslot::has_slots<> {
   // Transport flows: even is RTP, odd is RTCP
   std::map<int, mozilla::RefPtr<mozilla::TransportFlow> > mTransportFlows;
 
-  // Conduits: even is receive, odd is transmit (for easier correlation with
-  // flows)
-  std::map<int, mozilla::RefPtr<mozilla::AudioSessionConduit> > mAudioConduits;
-
   NS_INLINE_DECL_THREADSAFE_REFCOUNTING(PeerConnectionMedia)
 };
 

media/webrtc/signaling/test/mediaconduit_unittests.cpp

@@ -490,11 +490,11 @@ class TransportConduitTest : public ::testing::Test
   {
     //get pointer to AudioSessionConduit
     int err=0;
-    mAudioSession = mozilla::AudioSessionConduit::Create(NULL);
+    mAudioSession = mozilla::AudioSessionConduit::Create();
     if( !mAudioSession )
       ASSERT_NE(mAudioSession, (void*)NULL);
 
-    mAudioSession2 = mozilla::AudioSessionConduit::Create(NULL);
+    mAudioSession2 = mozilla::AudioSessionConduit::Create();
     if( !mAudioSession2 )
       ASSERT_NE(mAudioSession2, (void*)NULL);
 

media/webrtc/signaling/test/mediapipeline_unittest.cpp

@@ -48,7 +48,7 @@ class TestAgent {
       audio_prsock_(new TransportLayerPrsock()),
       audio_dtls_(new TransportLayerDtls()),
       audio_config_(109, "opus", 48000, 480, 1, 64000),
-      audio_conduit_(mozilla::AudioSessionConduit::Create(NULL)),
+      audio_conduit_(mozilla::AudioSessionConduit::Create()),
       audio_(),
       audio_pipeline_(),
       video_flow_(new TransportFlow()),

media/webrtc/webrtc_config.gypi

@@ -3,9 +3,6 @@
 # file, You can obtain one at http://mozilla.org/MPL/2.0/.
 
 # definitions to control what gets built in webrtc
-# NOTE!!! if you change something here, due to .gyp files not
-# being reprocessed on .gypi changes, run this before building:
-# "find . -name '*.gyp' | xargs touch"
 {
   'variables': {
     # basic stuff for everything
@@ -16,8 +13,6 @@
     'include_tests': 0,
     'use_system_libjpeg': 1,
     'use_system_libvpx': 1,
-# Creates AEC internal sample dump files in current directory
-#    'aec_debug_dump': 1,
 
     # codec enable/disables:
     # Note: if you change one here, you must modify shared_libs.mk!

modules/libpref/src/init/all.js

@@ -178,15 +178,6 @@ pref("media.gstreamer.enabled", true);
 pref("media.navigator.enabled", true);
 pref("media.peerconnection.enabled", false);
 pref("media.navigator.permission.disabled", false);
-// These values (aec, agc, and noice) are from media/webrtc/trunk/webrtc/common_types.h
-// kXxxUnchanged = 0, kXxxDefault = 1, and higher values are specific to each 
-// setting (for Xxx = Ec, Agc, or Ns).  Defaults are all set to kXxxDefault here.
-pref("media.peerconnection.aec_enabled", true);
-pref("media.peerconnection.aec", 1);
-pref("media.peerconnection.agc_enabled", false);
-pref("media.peerconnection.agc", 1);
-pref("media.peerconnection.noise_enabled", false);
-pref("media.peerconnection.noise", 1);
 #else
 #ifdef ANDROID
 pref("media.navigator.enabled", true);