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::EnumerateVideoDevices(MediaSourceType aMediaSource,
nsTArray<nsRefPtr<MediaEngineVideoSource> >* aVSources)
{
// We spawn threads to handle gUM runnables, so we must protect the member vars
MutexAutoLock lock(mMutex);
#ifdef MOZ_B2G_CAMERA
if (aMediaSource != MediaSourceType::Camera) {
// only supports camera sources
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 = 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, aMediaSource);
mVideoSources.Put(uuid, vSource); // Hashtable takes ownership.
aVSources->AppendElement(vSource);
}
}
return;
#else
ScopedCustomReleasePtr<webrtc::ViEBase> ptrViEBase;
ScopedCustomReleasePtr<webrtc::ViECapture> ptrViECapture;
webrtc::Config configSet;
webrtc::VideoEngine *videoEngine = nullptr;
bool *videoEngineInit = nullptr;
#ifdef MOZ_WIDGET_ANDROID
// get the JVM
JavaVM *jvm = mozilla::AndroidBridge::Bridge()->GetVM();
if (webrtc::VideoEngine::SetAndroidObjects(jvm) != 0) {
LOG(("VieCapture:SetAndroidObjects Failed"));
return;
}
#endif
switch (aMediaSource) {
case MediaSourceType::Window:
mWinEngineConfig.Set<webrtc::CaptureDeviceInfo>(
new webrtc::CaptureDeviceInfo(webrtc::CaptureDeviceType::Window));
if (!mWinEngine) {
if (!(mWinEngine = webrtc::VideoEngine::Create(mWinEngineConfig))) {
return;
}
}
videoEngine = mWinEngine;
videoEngineInit = &mWinEngineInit;
break;
case MediaSourceType::Application:
mAppEngineConfig.Set<webrtc::CaptureDeviceInfo>(
new webrtc::CaptureDeviceInfo(webrtc::CaptureDeviceType::Application));
if (!mAppEngine) {
if (!(mAppEngine = webrtc::VideoEngine::Create(mAppEngineConfig))) {
return;
}
}
videoEngine = mAppEngine;
videoEngineInit = &mAppEngineInit;
break;
case MediaSourceType::Screen:
mScreenEngineConfig.Set<webrtc::CaptureDeviceInfo>(
new webrtc::CaptureDeviceInfo(webrtc::CaptureDeviceType::Screen));
if (!mScreenEngine) {
if (!(mScreenEngine = webrtc::VideoEngine::Create(mScreenEngineConfig))) {
return;
}
}
videoEngine = mScreenEngine;
videoEngineInit = &mScreenEngineInit;
break;
case MediaSourceType::Browser:
mBrowserEngineConfig.Set<webrtc::CaptureDeviceInfo>(
new webrtc::CaptureDeviceInfo(webrtc::CaptureDeviceType::Browser));
if (!mBrowserEngine) {
if (!(mBrowserEngine = webrtc::VideoEngine::Create(mBrowserEngineConfig))) {
return;
}
}
videoEngine = mBrowserEngine;
videoEngineInit = &mBrowserEngineInit;
break;
case MediaSourceType::Camera:
// fall through
default:
if (!mVideoEngine) {
if (!(mVideoEngine = webrtc::VideoEngine::Create())) {
return;
}
}
videoEngine = mVideoEngine;
videoEngineInit = &mVideoEngineInit;
break;
}
ptrViEBase = webrtc::ViEBase::GetInterface(videoEngine);
if (!ptrViEBase) {
return;
}
if (ptrViEBase->Init() < 0) {
return;
}
*videoEngineInit = true;
ptrViECapture = webrtc::ViECapture::GetInterface(videoEngine);
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++) {
char deviceName[MediaEngineSource::kMaxDeviceNameLength];
char uniqueId[MediaEngineSource::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,
MediaEngineSource::kMaxUniqueIdLength);
LOG(("Number of Capabilities %d", numCaps));
for (int j = 0; j < numCaps; j++) {
if (ptrViECapture->GetCaptureCapability(uniqueId,
MediaEngineSource::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 refresh and append.
vSource->Refresh(i);
aVSources->AppendElement(vSource.get());
} else {
vSource = new MediaEngineWebRTCVideoSource(videoEngine, i, aMediaSource);
mVideoSources.Put(uuid, vSource); // Hashtable takes ownership.
aVSources->AppendElement(vSource);
}
}
if (mHasTabVideoSource || MediaSourceType::Browser == aMediaSource)
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::EnumerateAudioDevices(MediaSourceType aMediaSource,
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;
}
}
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);
int i;
#if defined(MOZ_WIDGET_ANDROID) || defined(MOZ_WIDGET_GONK)
i = 0; // Bug 1037025 - let the OS handle defaulting for now on android/b2g
#else
// -1 is "default communications device" depending on OS in webrtc.org code
i = -1;
#endif
for (; 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(
mThread, mVoiceEngine, i, deviceName, uniqueId
);
mAudioSources.Put(uuid, aSource); // Hashtable takes ownership.
aASources->AppendElement(aSource);
}
}
}