The Tor Browser: content/media/webrtc/MediaEngineWebRTC.cpp@44a2da4a2ab2 (annotated)

content/media/webrtc/MediaEngineWebRTC.cpp@44a2da4a2ab2 (annotated)

content/media/webrtc/MediaEngineWebRTC.cpp

Fri, 16 Jan 2015 04:50:19 +0100

author: Michael Schloh von Bennewitz <michael@schloh.com>
date: Fri, 16 Jan 2015 04:50:19 +0100
branch: TOR_BUG_9701
changeset 13: 44a2da4a2ab2
permissions: -rw-r--r--

Replace accessor implementation with direct member state manipulation, by
request https://trac.torproject.org/projects/tor/ticket/9701#comment:32

 /* This Source Code Form is subject to the terms of the Mozilla Public
  * License, v. 2.0. If a copy of the MPL was not distributed with this file,
  * You can obtain one at http://mozilla.org/MPL/2.0/. */
 #ifdef MOZ_LOGGING
 #define FORCE_PR_LOG
 #endif
 #if defined(PR_LOG)
 #error "This file must be #included before any IPDL-generated files or other files that #include prlog.h"
 #endif
 #include "nsIPrefService.h"
 #include "nsIPrefBranch.h"
 #include "CSFLog.h"
 #include "prenv.h"
 #ifdef PR_LOGGING
 static PRLogModuleInfo*
 GetUserMediaLog()
 {
   static PRLogModuleInfo *sLog;
   if (!sLog)
     sLog = PR_NewLogModule("GetUserMedia");
   return sLog;
 }
 #endif
 #include "MediaEngineWebRTC.h"
 #include "ImageContainer.h"
 #include "nsIComponentRegistrar.h"
 #include "MediaEngineTabVideoSource.h"
 #include "nsITabSource.h"
 #include "MediaTrackConstraints.h"
 #ifdef MOZ_WIDGET_ANDROID
 #include "AndroidJNIWrapper.h"
 #include "AndroidBridge.h"
 #endif
 #undef LOG
 #define LOG(args) PR_LOG(GetUserMediaLog(), PR_LOG_DEBUG, args)
 namespace mozilla {
 MediaEngineWebRTC::MediaEngineWebRTC(MediaEnginePrefs &aPrefs)
   : mMutex("mozilla::MediaEngineWebRTC")
   , mVideoEngine(nullptr)
   , mVoiceEngine(nullptr)
   , mVideoEngineInit(false)
   , mAudioEngineInit(false)
   , mHasTabVideoSource(false)
 {
 #ifndef MOZ_B2G_CAMERA
   nsCOMPtr<nsIComponentRegistrar> compMgr;
   NS_GetComponentRegistrar(getter_AddRefs(compMgr));
   if (compMgr) {
     compMgr->IsContractIDRegistered(NS_TABSOURCESERVICE_CONTRACTID, &mHasTabVideoSource);
   }
 #else
   AsyncLatencyLogger::Get()->AddRef();
 #endif
   // XXX
   gFarendObserver = new AudioOutputObserver();
 }
 void
 MediaEngineWebRTC::EnumerateVideoDevices(nsTArray<nsRefPtr<MediaEngineVideoSource> >* aVSources)
 {
 #ifdef MOZ_B2G_CAMERA
   MutexAutoLock lock(mMutex);
   /**
    * We still enumerate every time, in case a new device was plugged in since
    * the last call. TODO: Verify that WebRTC actually does deal with hotplugging
    * new devices (with or without new engine creation) and accordingly adjust.
    * Enumeration is not neccessary if GIPS reports the same set of devices
    * for a given instance of the engine. Likewise, if a device was plugged out,
    * mVideoSources must be updated.
    */
   int num = 0;
   nsresult result;
   result = ICameraControl::GetNumberOfCameras(num);
   if (num <= 0 || result != NS_OK) {
     return;
   }
   for (int i = 0; i < num; i++) {
     nsCString cameraName;
     result = ICameraControl::GetCameraName(i, cameraName);
     if (result != NS_OK) {
       continue;
     }
     nsRefPtr<MediaEngineWebRTCVideoSource> vSource;
     NS_ConvertUTF8toUTF16 uuid(cameraName);
     if (mVideoSources.Get(uuid, getter_AddRefs(vSource))) {
       // We've already seen this device, just append.
       aVSources->AppendElement(vSource.get());
     } else {
       vSource = new MediaEngineWebRTCVideoSource(i);
       mVideoSources.Put(uuid, vSource); // Hashtable takes ownership.
       aVSources->AppendElement(vSource);
     }
   }
   return;
 #else
   ScopedCustomReleasePtr<webrtc::ViEBase> ptrViEBase;
   ScopedCustomReleasePtr<webrtc::ViECapture> ptrViECapture;
   // We spawn threads to handle gUM runnables, so we must protect the member vars
   MutexAutoLock lock(mMutex);
 #ifdef MOZ_WIDGET_ANDROID
   jobject context = mozilla::AndroidBridge::Bridge()->GetGlobalContextRef();
   // get the JVM
   JavaVM *jvm = mozilla::AndroidBridge::Bridge()->GetVM();
   if (webrtc::VideoEngine::SetAndroidObjects(jvm, (void*)context) != 0) {
     LOG(("VieCapture:SetAndroidObjects Failed"));
     return;
   }
 #endif
   if (!mVideoEngine) {
     if (!(mVideoEngine = webrtc::VideoEngine::Create())) {
       return;
     }
   }
   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";
     }
     LOG(("%s Logging webrtc to %s level %d", __FUNCTION__, file, logs->level));
     mVideoEngine->SetTraceFilter(logs->level);
     mVideoEngine->SetTraceFile(file);
   }
   ptrViEBase = webrtc::ViEBase::GetInterface(mVideoEngine);
   if (!ptrViEBase) {
     return;
   }
   if (!mVideoEngineInit) {
     if (ptrViEBase->Init() < 0) {
       return;
     }
     mVideoEngineInit = true;
   }
   ptrViECapture = webrtc::ViECapture::GetInterface(mVideoEngine);
   if (!ptrViECapture) {
     return;
   }
   /**
    * We still enumerate every time, in case a new device was plugged in since
    * the last call. TODO: Verify that WebRTC actually does deal with hotplugging
    * new devices (with or without new engine creation) and accordingly adjust.
    * Enumeration is not neccessary if GIPS reports the same set of devices
    * for a given instance of the engine. Likewise, if a device was plugged out,
    * mVideoSources must be updated.
    */
   int num = ptrViECapture->NumberOfCaptureDevices();
   if (num <= 0) {
     return;
   }
   for (int i = 0; i < num; i++) {
     const unsigned int kMaxDeviceNameLength = 128; // XXX FIX!
     const unsigned int kMaxUniqueIdLength = 256;
     char deviceName[kMaxDeviceNameLength];
     char uniqueId[kMaxUniqueIdLength];
     // paranoia
     deviceName[0] = '\0';
     uniqueId[0] = '\0';
     int error = ptrViECapture->GetCaptureDevice(i, deviceName,
                                                 sizeof(deviceName), uniqueId,
                                                 sizeof(uniqueId));
     if (error) {
       LOG((" VieCapture:GetCaptureDevice: Failed %d",
            ptrViEBase->LastError() ));
       continue;
     }
 #ifdef DEBUG
     LOG(("  Capture Device Index %d, Name %s", i, deviceName));
     webrtc::CaptureCapability cap;
     int numCaps = ptrViECapture->NumberOfCapabilities(uniqueId, kMaxUniqueIdLength);
     LOG(("Number of Capabilities %d", numCaps));
     for (int j = 0; j < numCaps; j++) {
       if (ptrViECapture->GetCaptureCapability(uniqueId, kMaxUniqueIdLength,
                                               j, cap ) != 0 ) {
         break;
       }
       LOG(("type=%d width=%d height=%d maxFPS=%d",
            cap.rawType, cap.width, cap.height, cap.maxFPS ));
     }
 #endif
     if (uniqueId[0] == '\0') {
       // In case a device doesn't set uniqueId!
       strncpy(uniqueId, deviceName, sizeof(uniqueId));
       uniqueId[sizeof(uniqueId)-1] = '\0'; // strncpy isn't safe
     }
     nsRefPtr<MediaEngineWebRTCVideoSource> vSource;
     NS_ConvertUTF8toUTF16 uuid(uniqueId);
     if (mVideoSources.Get(uuid, getter_AddRefs(vSource))) {
       // We've already seen this device, just append.
       aVSources->AppendElement(vSource.get());
     } else {
       vSource = new MediaEngineWebRTCVideoSource(mVideoEngine, i);
       mVideoSources.Put(uuid, vSource); // Hashtable takes ownership.
       aVSources->AppendElement(vSource);
     }
   }
   if (mHasTabVideoSource)
     aVSources->AppendElement(new MediaEngineTabVideoSource());
   return;
 #endif
 }
 void
 MediaEngineWebRTC::EnumerateAudioDevices(nsTArray<nsRefPtr<MediaEngineAudioSource> >* aASources)
 {
   ScopedCustomReleasePtr<webrtc::VoEBase> ptrVoEBase;
   ScopedCustomReleasePtr<webrtc::VoEHardware> ptrVoEHw;
   // We spawn threads to handle gUM runnables, so we must protect the member vars
   MutexAutoLock lock(mMutex);
 #ifdef MOZ_WIDGET_ANDROID
   jobject context = mozilla::AndroidBridge::Bridge()->GetGlobalContextRef();
   // get the JVM
   JavaVM *jvm = mozilla::AndroidBridge::Bridge()->GetVM();
   JNIEnv *env = GetJNIForThread();
   if (webrtc::VoiceEngine::SetAndroidObjects(jvm, env, (void*)context) != 0) {
     LOG(("VoiceEngine:SetAndroidObjects Failed"));
     return;
   }
 #endif
   if (!mVoiceEngine) {
     mVoiceEngine = webrtc::VoiceEngine::Create();
     if (!mVoiceEngine) {
       return;
     }
   }
   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";
     }
     LOG(("Logging webrtc to %s level %d", __FUNCTION__, file, logs->level));
     mVoiceEngine->SetTraceFilter(logs->level);
     mVoiceEngine->SetTraceFile(file);
   }
   ptrVoEBase = webrtc::VoEBase::GetInterface(mVoiceEngine);
   if (!ptrVoEBase) {
     return;
   }
   if (!mAudioEngineInit) {
     if (ptrVoEBase->Init() < 0) {
       return;
     }
     mAudioEngineInit = true;
   }
   ptrVoEHw = webrtc::VoEHardware::GetInterface(mVoiceEngine);
   if (!ptrVoEHw)  {
     return;
   }
   int nDevices = 0;
   ptrVoEHw->GetNumOfRecordingDevices(nDevices);
   for (int i = 0; i < nDevices; i++) {
     // We use constants here because GetRecordingDeviceName takes char[128].
     char deviceName[128];
     char uniqueId[128];
     // paranoia; jingle doesn't bother with this
     deviceName[0] = '\0';
     uniqueId[0] = '\0';
     int error = ptrVoEHw->GetRecordingDeviceName(i, deviceName, uniqueId);
     if (error) {
       LOG((" VoEHardware:GetRecordingDeviceName: Failed %d",
            ptrVoEBase->LastError() ));
       continue;
     }
     if (uniqueId[0] == '\0') {
       // Mac and Linux don't set uniqueId!
       MOZ_ASSERT(sizeof(deviceName) == sizeof(uniqueId)); // total paranoia
       strcpy(uniqueId,deviceName); // safe given assert and initialization/error-check
     }
     nsRefPtr<MediaEngineWebRTCAudioSource> aSource;
     NS_ConvertUTF8toUTF16 uuid(uniqueId);
     if (mAudioSources.Get(uuid, getter_AddRefs(aSource))) {
       // We've already seen this device, just append.
       aASources->AppendElement(aSource.get());
     } else {
       aSource = new MediaEngineWebRTCAudioSource(
         mVoiceEngine, i, deviceName, uniqueId
       );
       mAudioSources.Put(uuid, aSource); // Hashtable takes ownership.
       aASources->AppendElement(aSource);
     }
   }
 }
 void
 MediaEngineWebRTC::Shutdown()
 {
   // This is likely paranoia
   MutexAutoLock lock(mMutex);
   if (mVideoEngine) {
     mVideoSources.Clear();
     webrtc::VideoEngine::Delete(mVideoEngine);
   }
   if (mVoiceEngine) {
     mAudioSources.Clear();
     webrtc::VoiceEngine::Delete(mVoiceEngine);
   }
   mVideoEngine = nullptr;
   mVoiceEngine = nullptr;
 }
 }

The Tor Browser / annotate

content/media/webrtc/MediaEngineWebRTC.cpp@44a2da4a2ab2 (annotated)

content/media/webrtc/MediaEngineWebRTC.cpp