/**
* Peforms intialization of the MANDATORY components of the Video Engine
*/
MediaConduitErrorCode WebrtcVideoConduit::Init()
{
CSFLogDebug(logTag, "%s ", __FUNCTION__);
#ifdef MOZ_WIDGET_ANDROID
jobject context = jsjni_GetGlobalContextRef();
// get the JVM
JavaVM *jvm = jsjni_GetVM();
JNIEnv* env;
if (jvm->GetEnv((void**)&env, JNI_VERSION_1_4) != JNI_OK) {
CSFLogError(logTag, "%s: could not get Java environment", __FUNCTION__);
return kMediaConduitSessionNotInited;
}
jvm->AttachCurrentThread(&env, nullptr);
if (webrtc::VideoEngine::SetAndroidObjects(jvm, (void*)context) != 0) {
CSFLogError(logTag, "%s: could not set Android objects", __FUNCTION__);
return kMediaConduitSessionNotInited;
}
env->DeleteGlobalRef(context);
#endif
if( !(mVideoEngine = webrtc::VideoEngine::Create()) )
{
CSFLogError(logTag, "%s Unable to create video engine ", __FUNCTION__);
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);
mVideoEngine->SetTraceFilter(logs->level);
mVideoEngine->SetTraceFile(file);
}
if( !(mPtrViEBase = ViEBase::GetInterface(mVideoEngine)))
{
CSFLogError(logTag, "%s Unable to get video base interface ", __FUNCTION__);
return kMediaConduitSessionNotInited;
}
if( !(mPtrViECapture = ViECapture::GetInterface(mVideoEngine)))
{
CSFLogError(logTag, "%s Unable to get video capture interface", __FUNCTION__);
return kMediaConduitSessionNotInited;
}
if( !(mPtrViECodec = ViECodec::GetInterface(mVideoEngine)))
{
CSFLogError(logTag, "%s Unable to get video codec interface ", __FUNCTION__);
return kMediaConduitSessionNotInited;
}
if( !(mPtrViENetwork = ViENetwork::GetInterface(mVideoEngine)))
{
CSFLogError(logTag, "%s Unable to get video network interface ", __FUNCTION__);
return kMediaConduitSessionNotInited;
}
if( !(mPtrViERender = ViERender::GetInterface(mVideoEngine)))
{
CSFLogError(logTag, "%s Unable to get video render interface ", __FUNCTION__);
return kMediaConduitSessionNotInited;
}
if( !(mPtrRTP = webrtc::ViERTP_RTCP::GetInterface(mVideoEngine)))
{
CSFLogError(logTag, "%s Unable to get video RTCP interface ", __FUNCTION__);
return kMediaConduitSessionNotInited;
}
CSFLogDebug(logTag, "%s Engine Created: Init'ng the interfaces ",__FUNCTION__);
if(mPtrViEBase->Init() == -1)
{
CSFLogError(logTag, " %s Video Engine Init Failed %d ",__FUNCTION__,
mPtrViEBase->LastError());
return kMediaConduitSessionNotInited;
}
if(mPtrViEBase->CreateChannel(mChannel) == -1)
{
CSFLogError(logTag, " %s Channel creation Failed %d ",__FUNCTION__,
mPtrViEBase->LastError());
return kMediaConduitChannelError;
}
if(mPtrViENetwork->RegisterSendTransport(mChannel, *this) == -1)
{
CSFLogError(logTag, "%s ViENetwork Failed %d ", __FUNCTION__,
mPtrViEBase->LastError());
return kMediaConduitTransportRegistrationFail;
}
mPtrExtCapture = 0;
if(mPtrViECapture->AllocateExternalCaptureDevice(mCapId,
mPtrExtCapture) == -1)
{
CSFLogError(logTag, "%s Unable to Allocate capture module: %d ",
__FUNCTION__, mPtrViEBase->LastError());
return kMediaConduitCaptureError;
}
if(mPtrViECapture->ConnectCaptureDevice(mCapId,mChannel) == -1)
{
CSFLogError(logTag, "%s Unable to Connect capture module: %d ",
__FUNCTION__,mPtrViEBase->LastError());
return kMediaConduitCaptureError;
}
if(mPtrViERender->AddRenderer(mChannel,
webrtc::kVideoI420,
(webrtc::ExternalRenderer*) this) == -1)
{
CSFLogError(logTag, "%s Failed to added external renderer ", __FUNCTION__);
return kMediaConduitInvalidRenderer;
}
// Set up some parameters, per juberti. Set MTU.
if(mPtrViENetwork->SetMTU(mChannel, 1200) != 0)
{
CSFLogError(logTag, "%s MTU Failed %d ", __FUNCTION__,
mPtrViEBase->LastError());
return kMediaConduitMTUError;
}
// Turn on RTCP and loss feedback reporting.
if(mPtrRTP->SetRTCPStatus(mChannel, webrtc::kRtcpCompound_RFC4585) != 0)
{
CSFLogError(logTag, "%s RTCPStatus Failed %d ", __FUNCTION__,
mPtrViEBase->LastError());
return kMediaConduitRTCPStatusError;
}
// Enable pli as key frame request method.
if(mPtrRTP->SetKeyFrameRequestMethod(mChannel,
webrtc::kViEKeyFrameRequestPliRtcp) != 0)
{
CSFLogError(logTag, "%s KeyFrameRequest Failed %d ", __FUNCTION__,
mPtrViEBase->LastError());
return kMediaConduitKeyFrameRequestError;
}
// Enable lossless transport
// XXX Note: We may want to disable this or limit it
if (mPtrRTP->SetNACKStatus(mChannel, true) != 0)
{
CSFLogError(logTag, "%s NACKStatus Failed %d ", __FUNCTION__,
mPtrViEBase->LastError());
return kMediaConduitNACKStatusError;
}
CSFLogError(logTag, "%s Initialization Done", __FUNCTION__);
return kMediaConduitNoError;
}
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
webrtc::ViEBase* ptrViEBase;
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 (mHasTabVideoSource)
aVSources->AppendElement(new MediaEngineTabVideoSource());
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);
}
}
ptrViEBase->Release();
ptrViECapture->Release();
return;
#endif
}
void
MediaEngineWebRTC::EnumerateAudioDevices(nsTArray<nsRefPtr<MediaEngineAudioSource> >* aASources)
{
webrtc::VoEBase* ptrVoEBase = nullptr;
webrtc::VoEHardware* ptrVoEHw = nullptr;
// 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);
}
}
ptrVoEHw->Release();
ptrVoEBase->Release();
}
/*
* WebRTCAudioConduit Implementation
*/
MediaConduitErrorCode WebrtcAudioConduit::Init(WebrtcAudioConduit *other)
{
CSFLogDebug(logTag, "%s this=%p other=%p", __FUNCTION__, this, other);
if (other) {
MOZ_ASSERT(!other->mOtherDirection);
other->mOtherDirection = this;
mOtherDirection = other;
// only one can call ::Create()/GetVoiceEngine()
MOZ_ASSERT(other->mVoiceEngine);
mVoiceEngine = other->mVoiceEngine;
} else {
//Per WebRTC APIs below function calls return NULL on failure
if(!(mVoiceEngine = webrtc::VoiceEngine::Create()))
{
CSFLogError(logTag, "%s Unable to create voice engine", __FUNCTION__);
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);
}
}
if(!(mPtrVoEBase = VoEBase::GetInterface(mVoiceEngine)))
{
CSFLogError(logTag, "%s Unable to initialize VoEBase", __FUNCTION__);
return kMediaConduitSessionNotInited;
}
if(!(mPtrVoENetwork = VoENetwork::GetInterface(mVoiceEngine)))
{
CSFLogError(logTag, "%s Unable to initialize VoENetwork", __FUNCTION__);
return kMediaConduitSessionNotInited;
}
if(!(mPtrVoECodec = VoECodec::GetInterface(mVoiceEngine)))
{
CSFLogError(logTag, "%s Unable to initialize VoEBCodec", __FUNCTION__);
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) {
mChannel = other->mChannel;
} else {
// init the engine with our audio device layer
if(mPtrVoEBase->Init() == -1)
{
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);
}
return kMediaConduitNoError;
}
/**
* Peforms intialization of the MANDATORY components of the Video Engine
*/
MediaConduitErrorCode WebrtcVideoConduit::Init(WebrtcVideoConduit *other)
{
CSFLogDebug(logTag, "%s this=%p other=%p", __FUNCTION__, this, other);
if (other) {
MOZ_ASSERT(!other->mOtherDirection);
other->mOtherDirection = this;
mOtherDirection = other;
// only one can call ::Create()/GetVideoEngine()
MOZ_ASSERT(other->mVideoEngine);
mVideoEngine = other->mVideoEngine;
} else {
#ifdef MOZ_WIDGET_ANDROID
jobject context = jsjni_GetGlobalContextRef();
// get the JVM
JavaVM *jvm = jsjni_GetVM();
if (webrtc::VideoEngine::SetAndroidObjects(jvm, (void*)context) != 0) {
CSFLogError(logTag, "%s: could not set Android objects", __FUNCTION__);
return kMediaConduitSessionNotInited;
}
#endif
// Per WebRTC APIs below function calls return nullptr on failure
if( !(mVideoEngine = webrtc::VideoEngine::Create()) )
{
CSFLogError(logTag, "%s Unable to create video engine ", __FUNCTION__);
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);
mVideoEngine->SetTraceFilter(logs->level);
mVideoEngine->SetTraceFile(file);
}
}
if( !(mPtrViEBase = ViEBase::GetInterface(mVideoEngine)))
{
CSFLogError(logTag, "%s Unable to get video base interface ", __FUNCTION__);
return kMediaConduitSessionNotInited;
}
if( !(mPtrViECapture = ViECapture::GetInterface(mVideoEngine)))
{
CSFLogError(logTag, "%s Unable to get video capture interface", __FUNCTION__);
return kMediaConduitSessionNotInited;
}
if( !(mPtrViECodec = ViECodec::GetInterface(mVideoEngine)))
{
CSFLogError(logTag, "%s Unable to get video codec interface ", __FUNCTION__);
return kMediaConduitSessionNotInited;
}
if( !(mPtrViENetwork = ViENetwork::GetInterface(mVideoEngine)))
{
CSFLogError(logTag, "%s Unable to get video network interface ", __FUNCTION__);
return kMediaConduitSessionNotInited;
}
if( !(mPtrViERender = ViERender::GetInterface(mVideoEngine)))
{
CSFLogError(logTag, "%s Unable to get video render interface ", __FUNCTION__);
return kMediaConduitSessionNotInited;
}
mPtrExtCodec = webrtc::ViEExternalCodec::GetInterface(mVideoEngine);
if (!mPtrExtCodec) {
CSFLogError(logTag, "%s Unable to get external codec interface: %d ",
__FUNCTION__,mPtrViEBase->LastError());
return kMediaConduitSessionNotInited;
}
if( !(mPtrRTP = webrtc::ViERTP_RTCP::GetInterface(mVideoEngine)))
{
CSFLogError(logTag, "%s Unable to get video RTCP interface ", __FUNCTION__);
return kMediaConduitSessionNotInited;
}
if (other) {
mChannel = other->mChannel;
mPtrExtCapture = other->mPtrExtCapture;
mCapId = other->mCapId;
} else {
CSFLogDebug(logTag, "%s Engine Created: Init'ng the interfaces ",__FUNCTION__);
if(mPtrViEBase->Init() == -1)
{
CSFLogError(logTag, " %s Video Engine Init Failed %d ",__FUNCTION__,
mPtrViEBase->LastError());
return kMediaConduitSessionNotInited;
}
if(mPtrViEBase->CreateChannel(mChannel) == -1)
{
CSFLogError(logTag, " %s Channel creation Failed %d ",__FUNCTION__,
mPtrViEBase->LastError());
return kMediaConduitChannelError;
}
if(mPtrViENetwork->RegisterSendTransport(mChannel, *this) == -1)
{
CSFLogError(logTag, "%s ViENetwork Failed %d ", __FUNCTION__,
mPtrViEBase->LastError());
return kMediaConduitTransportRegistrationFail;
}
if(mPtrViECapture->AllocateExternalCaptureDevice(mCapId,
mPtrExtCapture) == -1)
{
CSFLogError(logTag, "%s Unable to Allocate capture module: %d ",
__FUNCTION__, mPtrViEBase->LastError());
return kMediaConduitCaptureError;
}
if(mPtrViECapture->ConnectCaptureDevice(mCapId,mChannel) == -1)
{
CSFLogError(logTag, "%s Unable to Connect capture module: %d ",
__FUNCTION__,mPtrViEBase->LastError());
return kMediaConduitCaptureError;
}
if(mPtrViERender->AddRenderer(mChannel,
webrtc::kVideoI420,
(webrtc::ExternalRenderer*) this) == -1)
{
CSFLogError(logTag, "%s Failed to added external renderer ", __FUNCTION__);
return kMediaConduitInvalidRenderer;
}
// Set up some parameters, per juberti. Set MTU.
if(mPtrViENetwork->SetMTU(mChannel, 1200) != 0)
{
CSFLogError(logTag, "%s MTU Failed %d ", __FUNCTION__,
mPtrViEBase->LastError());
return kMediaConduitMTUError;
}
// Turn on RTCP and loss feedback reporting.
if(mPtrRTP->SetRTCPStatus(mChannel, webrtc::kRtcpCompound_RFC4585) != 0)
{
CSFLogError(logTag, "%s RTCPStatus Failed %d ", __FUNCTION__,
mPtrViEBase->LastError());
return kMediaConduitRTCPStatusError;
}
}
#ifdef VIDEOCONDUIT_INSERT_TIMESTAMP
mStartTime = PR_IntervalNow();
mSentFrames = 0;
#endif
CSFLogError(logTag, "%s Initialization Done", __FUNCTION__);
return kMediaConduitNoError;
}
/**
* Performs initialization of the MANDATORY components of the Video Engine
*/
MediaConduitErrorCode WebrtcVideoConduit::Init(WebrtcVideoConduit *other)
{
CSFLogDebug(logTag, "%s this=%p other=%p", __FUNCTION__, this, other);
#ifdef MOZILLA_INTERNAL_API
// already know we must be on MainThread barring unit test weirdness
MOZ_ASSERT(NS_IsMainThread());
nsresult rv;
nsCOMPtr<nsIPrefService> prefs = do_GetService("@mozilla.org/preferences-service;1", &rv);
if (!NS_WARN_IF(NS_FAILED(rv)))
{
nsCOMPtr<nsIPrefBranch> branch = do_QueryInterface(prefs);
if (branch)
{
int32_t temp;
NS_WARN_IF(NS_FAILED(branch->GetBoolPref("media.video.test_latency", &mVideoLatencyTestEnable)));
NS_WARN_IF(NS_FAILED(branch->GetIntPref("media.peerconnection.video.min_bitrate", &temp)));
if (temp >= 0) {
mMinBitrate = temp;
}
NS_WARN_IF(NS_FAILED(branch->GetIntPref("media.peerconnection.video.start_bitrate", &temp)));
if (temp >= 0) {
mStartBitrate = temp;
}
NS_WARN_IF(NS_FAILED(branch->GetIntPref("media.peerconnection.video.max_bitrate", &temp)));
if (temp >= 0) {
mMaxBitrate = temp;
}
}
}
#endif
if (other) {
MOZ_ASSERT(!other->mOtherDirection);
other->mOtherDirection = this;
mOtherDirection = other;
// only one can call ::Create()/GetVideoEngine()
MOZ_ASSERT(other->mVideoEngine);
mVideoEngine = other->mVideoEngine;
} else {
#ifdef MOZ_WIDGET_ANDROID
// get the JVM
JavaVM *jvm = jsjni_GetVM();
if (webrtc::VideoEngine::SetAndroidObjects(jvm) != 0) {
CSFLogError(logTag, "%s: could not set Android objects", __FUNCTION__);
return kMediaConduitSessionNotInited;
}
#endif
// Per WebRTC APIs below function calls return nullptr on failure
if( !(mVideoEngine = webrtc::VideoEngine::Create()) )
{
CSFLogError(logTag, "%s Unable to create video engine ", __FUNCTION__);
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);
mVideoEngine->SetTraceFilter(logs->level);
mVideoEngine->SetTraceFile(file);
}
}
if( !(mPtrViEBase = ViEBase::GetInterface(mVideoEngine)))
{
CSFLogError(logTag, "%s Unable to get video base interface ", __FUNCTION__);
return kMediaConduitSessionNotInited;
}
if( !(mPtrViECapture = ViECapture::GetInterface(mVideoEngine)))
{
CSFLogError(logTag, "%s Unable to get video capture interface", __FUNCTION__);
return kMediaConduitSessionNotInited;
}
if( !(mPtrViECodec = ViECodec::GetInterface(mVideoEngine)))
{
CSFLogError(logTag, "%s Unable to get video codec interface ", __FUNCTION__);
return kMediaConduitSessionNotInited;
}
if( !(mPtrViENetwork = ViENetwork::GetInterface(mVideoEngine)))
{
CSFLogError(logTag, "%s Unable to get video network interface ", __FUNCTION__);
return kMediaConduitSessionNotInited;
}
if( !(mPtrViERender = ViERender::GetInterface(mVideoEngine)))
{
CSFLogError(logTag, "%s Unable to get video render interface ", __FUNCTION__);
return kMediaConduitSessionNotInited;
}
if( !(mPtrRTP = webrtc::ViERTP_RTCP::GetInterface(mVideoEngine)))
{
CSFLogError(logTag, "%s Unable to get video RTCP interface ", __FUNCTION__);
return kMediaConduitSessionNotInited;
}
if ( !(mPtrExtCodec = webrtc::ViEExternalCodec::GetInterface(mVideoEngine)))
{
CSFLogError(logTag, "%s Unable to get external codec interface %d ",
__FUNCTION__, mPtrViEBase->LastError());
return kMediaConduitSessionNotInited;
}
if (other) {
mChannel = other->mChannel;
mPtrExtCapture = other->mPtrExtCapture;
mCapId = other->mCapId;
} else {
CSFLogDebug(logTag, "%s Engine Created: Init'ng the interfaces ",__FUNCTION__);
if(mPtrViEBase->Init() == -1)
{
CSFLogError(logTag, " %s Video Engine Init Failed %d ",__FUNCTION__,
mPtrViEBase->LastError());
return kMediaConduitSessionNotInited;
}
if(mPtrViEBase->CreateChannel(mChannel) == -1)
{
CSFLogError(logTag, " %s Channel creation Failed %d ",__FUNCTION__,
mPtrViEBase->LastError());
return kMediaConduitChannelError;
}
if(mPtrViENetwork->RegisterSendTransport(mChannel, *this) == -1)
{
CSFLogError(logTag, "%s ViENetwork Failed %d ", __FUNCTION__,
mPtrViEBase->LastError());
return kMediaConduitTransportRegistrationFail;
}
if(mPtrViECapture->AllocateExternalCaptureDevice(mCapId,
mPtrExtCapture) == -1)
{
CSFLogError(logTag, "%s Unable to Allocate capture module: %d ",
__FUNCTION__, mPtrViEBase->LastError());
return kMediaConduitCaptureError;
}
if(mPtrViECapture->ConnectCaptureDevice(mCapId,mChannel) == -1)
{
CSFLogError(logTag, "%s Unable to Connect capture module: %d ",
__FUNCTION__,mPtrViEBase->LastError());
return kMediaConduitCaptureError;
}
if(mPtrViERender->AddRenderer(mChannel,
webrtc::kVideoI420,
(webrtc::ExternalRenderer*) this) == -1)
{
CSFLogError(logTag, "%s Failed to added external renderer ", __FUNCTION__);
return kMediaConduitInvalidRenderer;
}
// Set up some parameters, per juberti. Set MTU.
if(mPtrViENetwork->SetMTU(mChannel, 1200) != 0)
{
CSFLogError(logTag, "%s MTU Failed %d ", __FUNCTION__,
mPtrViEBase->LastError());
return kMediaConduitMTUError;
}
// Turn on RTCP and loss feedback reporting.
if(mPtrRTP->SetRTCPStatus(mChannel, webrtc::kRtcpCompound_RFC4585) != 0)
{
CSFLogError(logTag, "%s RTCPStatus Failed %d ", __FUNCTION__,
mPtrViEBase->LastError());
return kMediaConduitRTCPStatusError;
}
}
CSFLogError(logTag, "%s Initialization Done", __FUNCTION__);
return kMediaConduitNoError;
}
