@ -10,6 +10,7 @@
# include <thread>
# include <utility>
# include "CallDevices.h"
# include "ChatPage.h"
# include "Logging.h"
# include "UserSettingsPage.h"
@ -29,14 +30,20 @@ extern "C"
// https://github.com/vector-im/riot-web/issues/10173
# define STUN_SERVER "stun: //turn.matrix.org:3478"
Q_DECLARE_METATYPE ( webrtc : : CallType )
Q_DECLARE_METATYPE ( webrtc : : State )
using webrtc : : CallType ;
using webrtc : : State ;
WebRTCSession : : WebRTCSession ( )
: QObject ( )
, devices_ ( CallDevices : : instance ( ) )
{
qRegisterMetaType < webrtc : : CallType > ( ) ;
qmlRegisterUncreatableMetaObject (
webrtc : : staticMetaObject , " im.nheko " , 1 , 0 , " CallType " , " Can't instantiate enum " ) ;
qRegisterMetaType < webrtc : : State > ( ) ;
qmlRegisterUncreatableMetaObject (
webrtc : : staticMetaObject , " im.nheko " , 1 , 0 , " WebRTCState " , " Can't instantiate enum " ) ;
@ -82,9 +89,10 @@ namespace {
std : : string localsdp_ ;
std : : vector < mtx : : events : : msg : : CallCandidates : : Candidate > localcandidates_ ;
bool haveAudioStream_ ;
bool haveVideoStream_ ;
GstPad * insetSinkPad_ = nullptr ;
bool haveAudioStream_ = false ;
bool haveVideoStream_ = false ;
GstPad * localPiPSinkPad_ = nullptr ;
GstPad * remotePiPSinkPad_ = nullptr ;
gboolean
newBusMessage ( GstBus * bus G_GNUC_UNUSED , GstMessage * msg , gpointer user_data )
@ -166,7 +174,6 @@ createAnswer(GstPromise *promise, gpointer webrtc)
g_signal_emit_by_name ( webrtc , " create-answer " , nullptr , promise ) ;
}
# if GST_CHECK_VERSION(1, 18, 0)
void
iceGatheringStateChanged ( GstElement * webrtc ,
GParamSpec * pspec G_GNUC_UNUSED ,
@ -186,23 +193,6 @@ iceGatheringStateChanged(GstElement *webrtc,
}
}
# else
gboolean
onICEGatheringCompletion ( gpointer timerid )
{
* ( guint * ) ( timerid ) = 0 ;
if ( WebRTCSession : : instance ( ) . isOffering ( ) ) {
emit WebRTCSession : : instance ( ) . offerCreated ( localsdp_ , localcandidates_ ) ;
emit WebRTCSession : : instance ( ) . stateChanged ( State : : OFFERSENT ) ;
} else {
emit WebRTCSession : : instance ( ) . answerCreated ( localsdp_ , localcandidates_ ) ;
emit WebRTCSession : : instance ( ) . stateChanged ( State : : ANSWERSENT ) ;
}
return FALSE ;
}
# endif
void
addLocalICECandidate ( GstElement * webrtc G_GNUC_UNUSED ,
guint mlineIndex ,
@ -210,28 +200,7 @@ addLocalICECandidate(GstElement *webrtc G_GNUC_UNUSED,
gpointer G_GNUC_UNUSED )
{
nhlog : : ui ( ) - > debug ( " WebRTC: local candidate: (m-line:{}):{} " , mlineIndex , candidate ) ;
# if GST_CHECK_VERSION(1, 18, 0)
localcandidates_ . push_back ( { std : : string ( ) /*max-bundle*/ , ( uint16_t ) mlineIndex , candidate } ) ;
return ;
# else
if ( WebRTCSession : : instance ( ) . state ( ) > = State : : OFFERSENT ) {
emit WebRTCSession : : instance ( ) . newICECandidate (
{ std : : string ( ) /*max-bundle*/ , ( uint16_t ) mlineIndex , candidate } ) ;
return ;
}
localcandidates_ . push_back ( { std : : string ( ) /*max-bundle*/ , ( uint16_t ) mlineIndex , candidate } ) ;
// GStreamer v1.16: webrtcbin's notify::ice-gathering-state triggers
// GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE too early. Fixed in v1.18.
// Use a 1s timeout in the meantime
static guint timerid = 0 ;
if ( timerid )
g_source_remove ( timerid ) ;
timerid = g_timeout_add ( 1000 , onICEGatheringCompletion , & timerid ) ;
# endif
}
void
@ -320,7 +289,6 @@ testPacketLoss(gpointer G_GNUC_UNUSED)
return FALSE ;
}
# if GST_CHECK_VERSION(1, 18, 0)
void
setWaitForKeyFrame ( GstBin * decodebin G_GNUC_UNUSED , GstElement * element , gpointer G_GNUC_UNUSED )
{
@ -329,7 +297,6 @@ setWaitForKeyFrame(GstBin *decodebin G_GNUC_UNUSED, GstElement *element, gpointe
" rtpvp8depay " ) )
g_object_set ( element , " wait-for-keyframe " , TRUE , nullptr ) ;
}
# endif
GstElement *
newAudioSinkChain ( GstElement * pipe )
@ -357,6 +324,7 @@ newVideoSinkChain(GstElement *pipe)
GstElement * glcolorconvert = gst_element_factory_make ( " glcolorconvert " , nullptr ) ;
GstElement * qmlglsink = gst_element_factory_make ( " qmlglsink " , nullptr ) ;
GstElement * glsinkbin = gst_element_factory_make ( " glsinkbin " , nullptr ) ;
g_object_set ( compositor , " background " , 1 , nullptr ) ;
g_object_set ( qmlglsink , " widget " , WebRTCSession : : instance ( ) . getVideoItem ( ) , nullptr ) ;
g_object_set ( glsinkbin , " sink " , qmlglsink , nullptr ) ;
gst_bin_add_many (
@ -382,44 +350,98 @@ getResolution(GstPad *pad)
return ret ;
}
std : : pair < int , int >
getResolution ( GstElement * pipe , const gchar * elementName , const gchar * padName )
{
GstElement * element = gst_bin_get_by_name ( GST_BIN ( pipe ) , elementName ) ;
GstPad * pad = gst_element_get_static_pad ( element , padName ) ;
auto ret = getResolution ( pad ) ;
gst_object_unref ( pad ) ;
gst_object_unref ( element ) ;
return ret ;
}
std : : pair < int , int >
getPiPDimensions ( const std : : pair < int , int > & resolution , int fullWidth , double scaleFactor )
{
int pipWidth = fullWidth * scaleFactor ;
int pipHeight = static_cast < double > ( resolution . second ) / resolution . first * pipWidth ;
return { pipWidth , pipHeight } ;
}
void
addCameraView ( GstElement * pipe , const std : : pair < int , int > & videoCallSize )
addLocalPiP ( GstElement * pipe , const std : : pair < int , int > & videoCallSize )
{
// embed localUser's camera into received video (CallType::VIDEO)
// OR embed screen share into received video (CallType::SCREEN)
GstElement * tee = gst_bin_get_by_name ( GST_BIN ( pipe ) , " videosrctee " ) ;
if ( ! tee )
return ;
GstElement * queue = gst_element_factory_make ( " queue " , nullptr ) ;
GstElement * videorate = gst_element_factory_make ( " videorate " , nullptr ) ;
gst_bin_add_many ( GST_BIN ( pipe ) , queue , videorate , nullptr ) ;
gst_element_link_many ( tee , queue , videorate , nullptr ) ;
GstElement * queue = gst_element_factory_make ( " queue " , nullptr ) ;
gst_bin_add ( GST_BIN ( pipe ) , queue ) ;
gst_element_link ( tee , queue ) ;
gst_element_sync_state_with_parent ( queue ) ;
gst_element_sync_state_with_parent ( videorate ) ;
gst_object_unref ( tee ) ;
GstElement * camerafilter = gst_bin_get_by_name ( GST_BIN ( pipe ) , " camerafilter " ) ;
GstPad * filtersinkpad = gst_element_get_static_pad ( camerafilter , " sink " ) ;
auto cameraResolution = getResolution ( filtersinkpad ) ;
int insetWidth = videoCallSize . first / 4 ;
int insetHeight =
static_cast < double > ( cameraResolution . second ) / cameraResolution . first * insetWidth ;
nhlog : : ui ( ) - > debug ( " WebRTC: picture-in-picture size: {}x{} " , insetWidth , insetHeight ) ;
gst_object_unref ( filtersinkpad ) ;
gst_object_unref ( camerafilter ) ;
GstPad * camerapad = gst_element_get_static_pad ( videorate , " src " ) ;
GstElement * compositor = gst_bin_get_by_name ( GST_BIN ( pipe ) , " compositor " ) ;
insetSinkPad_ = gst_element_get_request_pad ( compositor , " sink_%u " ) ;
g_object_set ( insetSinkPad_ , " zorder " , 2 , nullptr ) ;
g_object_set ( insetSinkPad_ , " width " , insetWidth , " height " , insetHeight , nullptr ) ;
localPiPSinkPad_ = gst_element_get_request_pad ( compositor , " sink_%u " ) ;
g_object_set ( localPiPSinkPad_ , " zorder " , 2 , nullptr ) ;
bool isVideo = WebRTCSession : : instance ( ) . callType ( ) = = CallType : : VIDEO ;
const gchar * element = isVideo ? " camerafilter " : " screenshare " ;
const gchar * pad = isVideo ? " sink " : " src " ;
auto resolution = getResolution ( pipe , element , pad ) ;
auto pipSize = getPiPDimensions ( resolution , videoCallSize . first , 0.25 ) ;
nhlog : : ui ( ) - > debug (
" WebRTC: local picture-in-picture: {}x{} " , pipSize . first , pipSize . second ) ;
g_object_set ( localPiPSinkPad_ , " width " , pipSize . first , " height " , pipSize . second , nullptr ) ;
gint offset = videoCallSize . first / 80 ;
g_object_set ( insetSinkPad_ , " xpos " , offset , " ypos " , offset , nullptr ) ;
if ( GST_PAD_LINK_FAILED ( gst_pad_link ( camerapad , insetSinkPad_ ) ) )
nhlog : : ui ( ) - > error ( " WebRTC: failed to link camera view chain " ) ;
gst_object_unref ( camerapad ) ;
g_object_set ( localPiPSinkPad_ , " xpos " , offset , " ypos " , offset , nullptr ) ;
GstPad * srcpad = gst_element_get_static_pad ( queue , " src " ) ;
if ( GST_PAD_LINK_FAILED ( gst_pad_link ( srcpad , localPiPSinkPad_ ) ) )
nhlog : : ui ( ) - > error ( " WebRTC: failed to link local PiP elements " ) ;
gst_object_unref ( srcpad ) ;
gst_object_unref ( compositor ) ;
}
void
addRemotePiP ( GstElement * pipe )
{
// embed localUser's camera into screen image being shared
if ( remotePiPSinkPad_ ) {
auto camRes = getResolution ( pipe , " camerafilter " , " sink " ) ;
auto shareRes = getResolution ( pipe , " screenshare " , " src " ) ;
auto pipSize = getPiPDimensions ( camRes , shareRes . first , 0.2 ) ;
nhlog : : ui ( ) - > debug (
" WebRTC: screen share picture-in-picture: {}x{} " , pipSize . first , pipSize . second ) ;
gint offset = shareRes . first / 100 ;
g_object_set ( remotePiPSinkPad_ , " zorder " , 2 , nullptr ) ;
g_object_set (
remotePiPSinkPad_ , " width " , pipSize . first , " height " , pipSize . second , nullptr ) ;
g_object_set ( remotePiPSinkPad_ ,
" xpos " ,
shareRes . first - pipSize . first - offset ,
" ypos " ,
shareRes . second - pipSize . second - offset ,
nullptr ) ;
}
}
void
addLocalVideo ( GstElement * pipe )
{
GstElement * queue = newVideoSinkChain ( pipe ) ;
GstElement * tee = gst_bin_get_by_name ( GST_BIN ( pipe ) , " videosrctee " ) ;
GstPad * srcpad = gst_element_get_request_pad ( tee , " src_%u " ) ;
GstPad * sinkpad = gst_element_get_static_pad ( queue , " sink " ) ;
if ( GST_PAD_LINK_FAILED ( gst_pad_link ( srcpad , sinkpad ) ) )
nhlog : : ui ( ) - > error ( " WebRTC: failed to link videosrctee -> video sink chain " ) ;
gst_object_unref ( srcpad ) ;
}
void
linkNewPad ( GstElement * decodebin , GstPad * newpad , GstElement * pipe )
{
@ -455,7 +477,7 @@ linkNewPad(GstElement *decodebin, GstPad *newpad, GstElement *pipe)
nhlog : : ui ( ) - > info ( " WebRTC: incoming video resolution: {}x{} " ,
videoCallSize . first ,
videoCallSize . second ) ;
addCameraView ( pipe , videoCallSize ) ;
addLocalPiP ( pipe , videoCallSize ) ;
} else {
g_free ( mediaType ) ;
nhlog : : ui ( ) - > error ( " WebRTC: unknown pad type: {} " , GST_PAD_NAME ( newpad ) ) ;
@ -467,7 +489,7 @@ linkNewPad(GstElement *decodebin, GstPad *newpad, GstElement *pipe)
if ( GST_PAD_LINK_FAILED ( gst_pad_link ( newpad , queuepad ) ) )
nhlog : : ui ( ) - > error ( " WebRTC: unable to link new pad " ) ;
else {
if ( ! session - > isVideo ( ) | |
if ( session - > callType ( ) = = CallType : : VOICE | |
( haveAudioStream_ & &
( haveVideoStream_ | | session - > isRemoteVideoRecvOnly ( ) ) ) ) {
emit session - > stateChanged ( State : : CONNECTED ) ;
@ -477,6 +499,9 @@ linkNewPad(GstElement *decodebin, GstPad *newpad, GstElement *pipe)
keyFrameRequestData_ . timerid =
g_timeout_add_seconds ( 3 , testPacketLoss , nullptr ) ;
}
addRemotePiP ( pipe ) ;
if ( session - > isRemoteVideoRecvOnly ( ) )
addLocalVideo ( pipe ) ;
}
}
gst_object_unref ( queuepad ) ;
@ -495,9 +520,7 @@ addDecodeBin(GstElement *webrtc G_GNUC_UNUSED, GstPad *newpad, GstElement *pipe)
// hardware decoding needs investigation; eg rendering fails if vaapi plugin installed
g_object_set ( decodebin , " force-sw-decoders " , TRUE , nullptr ) ;
g_signal_connect ( decodebin , " pad-added " , G_CALLBACK ( linkNewPad ) , pipe ) ;
# if GST_CHECK_VERSION(1, 18, 0)
g_signal_connect ( decodebin , " element-added " , G_CALLBACK ( setWaitForKeyFrame ) , nullptr ) ;
# endif
gst_bin_add ( GST_BIN ( pipe ) , decodebin ) ;
gst_element_sync_state_with_parent ( decodebin ) ;
GstPad * sinkpad = gst_element_get_static_pad ( decodebin , " sink " ) ;
@ -523,14 +546,17 @@ getMediaAttributes(const GstSDPMessage *sdp,
const char * mediaType ,
const char * encoding ,
int & payloadType ,
bool & recvOnly )
bool & recvOnly ,
bool & sendOnly )
{
payloadType = - 1 ;
recvOnly = false ;
sendOnly = false ;
for ( guint mlineIndex = 0 ; mlineIndex < gst_sdp_message_medias_len ( sdp ) ; + + mlineIndex ) {
const GstSDPMedia * media = gst_sdp_message_get_media ( sdp , mlineIndex ) ;
if ( ! std : : strcmp ( gst_sdp_media_get_media ( media ) , mediaType ) ) {
recvOnly = gst_sdp_media_get_attribute_val ( media , " recvonly " ) ! = nullptr ;
sendOnly = gst_sdp_media_get_attribute_val ( media , " sendonly " ) ! = nullptr ;
const gchar * rtpval = nullptr ;
for ( guint n = 0 ; n = = 0 | | rtpval ; + + n ) {
rtpval = gst_sdp_media_get_attribute_val_n ( media , " rtpmap " , n ) ;
@ -603,17 +629,12 @@ WebRTCSession::havePlugins(bool isVideo, std::string *errorMessage)
}
bool
WebRTCSession : : createOffer ( bool isVideo )
WebRTCSession : : createOffer ( CallType callType , uint32_t shareWindowId )
{
isOffering_ = true ;
isVideo_ = isVideo ;
isRemoteVideoRecvOnly_ = false ;
videoItem_ = nullptr ;
haveAudioStream_ = false ;
haveVideoStream_ = false ;
insetSinkPad_ = nullptr ;
localsdp_ . clear ( ) ;
localcandidates_ . clear ( ) ;
clear ( ) ;
isOffering_ = true ;
callType_ = callType ;
shareWindowId_ = shareWindowId ;
// opus and vp8 rtp payload types must be defined dynamically
// therefore from the range [96-127]
@ -630,22 +651,15 @@ WebRTCSession::acceptOffer(const std::string &sdp)
if ( state_ ! = State : : DISCONNECTED )
return false ;
isOffering_ = false ;
isRemoteVideoRecvOnly_ = false ;
videoItem_ = nullptr ;
haveAudioStream_ = false ;
haveVideoStream_ = false ;
insetSinkPad_ = nullptr ;
localsdp_ . clear ( ) ;
localcandidates_ . clear ( ) ;
clear ( ) ;
GstWebRTCSessionDescription * offer = parseSDP ( sdp , GST_WEBRTC_SDP_TYPE_OFFER ) ;
if ( ! offer )
return false ;
int opusPayloadType ;
bool recvOnly ;
if ( getMediaAttributes ( offer - > sdp , " audio " , " opus " , opusPayloadType , recvOnly ) ) {
bool sendOnly ;
if ( getMediaAttributes ( offer - > sdp , " audio " , " opus " , opusPayloadType , recvOnly , sendOnly ) ) {
if ( opusPayloadType = = - 1 ) {
nhlog : : ui ( ) - > error ( " WebRTC: remote audio offer - no opus encoding " ) ;
gst_webrtc_session_description_free ( offer ) ;
@ -658,13 +672,18 @@ WebRTCSession::acceptOffer(const std::string &sdp)
}
int vp8PayloadType ;
isVideo_ =
getMediaAttributes ( offer - > sdp , " video " , " vp8 " , vp8PayloadType , isRemoteVideoRecvOnly_ ) ;
if ( isVideo_ & & vp8PayloadType = = - 1 ) {
bool isVideo = getMediaAttributes ( offer - > sdp ,
" video " ,
" vp8 " ,
vp8PayloadType ,
isRemoteVideoRecvOnly_ ,
isRemoteVideoSendOnly_ ) ;
if ( isVideo & & vp8PayloadType = = - 1 ) {
nhlog : : ui ( ) - > error ( " WebRTC: remote video offer - no vp8 encoding " ) ;
gst_webrtc_session_description_free ( offer ) ;
return false ;
}
callType_ = isVideo ? CallType : : VIDEO : CallType : : VOICE ;
if ( ! startPipeline ( opusPayloadType , vp8PayloadType ) ) {
gst_webrtc_session_description_free ( offer ) ;
@ -695,10 +714,14 @@ WebRTCSession::acceptAnswer(const std::string &sdp)
return false ;
}
if ( isVideo_ ) {
if ( callType_ ! = CallType : : VOICE ) {
int unused ;
if ( ! getMediaAttributes (
answer - > sdp , " video " , " vp8 " , unused , isRemoteVideoRecvOnly_ ) )
if ( ! getMediaAttributes ( answer - > sdp ,
" video " ,
" vp8 " ,
unused ,
isRemoteVideoRecvOnly_ ,
isRemoteVideoSendOnly_ ) )
isRemoteVideoRecvOnly_ = true ;
}
@ -769,11 +792,10 @@ WebRTCSession::startPipeline(int opusPayloadType, int vp8PayloadType)
gst_element_set_state ( pipe_ , GST_STATE_READY ) ;
g_signal_connect ( webrtc_ , " pad-added " , G_CALLBACK ( addDecodeBin ) , pipe_ ) ;
# if GST_CHECK_VERSION(1, 18, 0)
// capture ICE gathering completion
g_signal_connect (
webrtc_ , " notify::ice-gathering-state " , G_CALLBACK ( iceGatheringStateChanged ) , nullptr ) ;
# endif
// webrtcbin lifetime is the same as that of the pipeline
gst_object_unref ( webrtc_ ) ;
@ -855,40 +877,115 @@ WebRTCSession::createPipeline(int opusPayloadType, int vp8PayloadType)
return false ;
}
return isVideo_ ? addVideoPipeline ( vp8PayloadType ) : true ;
return callType_ = = CallType : : VOICE | | isRemoteVideoSendOnly_
? true
: addVideoPipeline ( vp8PayloadType ) ;
}
bool
WebRTCSession : : addVideoPipeline ( int vp8PayloadType )
{
// allow incoming video calls despite localUser having no webcam
if ( ! devices_ . haveCamera ( ) )
if ( callType_ = = CallType : : VIDEO & & ! devices_ . haveCamera ( ) )
return ! isOffering_ ;
std : : pair < int , int > resolution ;
std : : pair < int , int > frameRate ;
GstDevice * device = devices_ . videoDevice ( resolution , frameRate ) ;
if ( ! device )
return false ;
GstElement * source = gst_device_create_element ( device , nullptr ) ;
auto settings = ChatPage : : instance ( ) - > userSettings ( ) ;
GstElement * camerafilter = nullptr ;
GstElement * videoconvert = gst_element_factory_make ( " videoconvert " , nullptr ) ;
GstElement * capsfilter = gst_element_factory_make ( " capsfilter " , " camerafilter " ) ;
GstCaps * caps = gst_caps_new_simple ( " video/x-raw " ,
" width " ,
G_TYPE_INT ,
resolution . first ,
" height " ,
G_TYPE_INT ,
resolution . second ,
" framerate " ,
GST_TYPE_FRACTION ,
frameRate . first ,
frameRate . second ,
nullptr ) ;
g_object_set ( capsfilter , " caps " , caps , nullptr ) ;
gst_caps_unref ( caps ) ;
GstElement * tee = gst_element_factory_make ( " tee " , " videosrctee " ) ;
GstElement * tee = gst_element_factory_make ( " tee " , " videosrctee " ) ;
gst_bin_add_many ( GST_BIN ( pipe_ ) , videoconvert , tee , nullptr ) ;
if ( callType_ = = CallType : : VIDEO | | ( settings - > screenSharePiP ( ) & & devices_ . haveCamera ( ) ) ) {
std : : pair < int , int > resolution ;
std : : pair < int , int > frameRate ;
GstDevice * device = devices_ . videoDevice ( resolution , frameRate ) ;
if ( ! device )
return false ;
GstElement * camera = gst_device_create_element ( device , nullptr ) ;
GstCaps * caps = gst_caps_new_simple ( " video/x-raw " ,
" width " ,
G_TYPE_INT ,
resolution . first ,
" height " ,
G_TYPE_INT ,
resolution . second ,
" framerate " ,
GST_TYPE_FRACTION ,
frameRate . first ,
frameRate . second ,
nullptr ) ;
camerafilter = gst_element_factory_make ( " capsfilter " , " camerafilter " ) ;
g_object_set ( camerafilter , " caps " , caps , nullptr ) ;
gst_caps_unref ( caps ) ;
gst_bin_add_many ( GST_BIN ( pipe_ ) , camera , camerafilter , nullptr ) ;
if ( ! gst_element_link_many ( camera , videoconvert , camerafilter , nullptr ) ) {
nhlog : : ui ( ) - > error ( " WebRTC: failed to link camera elements " ) ;
return false ;
}
if ( callType_ = = CallType : : VIDEO & & ! gst_element_link ( camerafilter , tee ) ) {
nhlog : : ui ( ) - > error ( " WebRTC: failed to link camerafilter -> tee " ) ;
return false ;
}
}
if ( callType_ = = CallType : : SCREEN ) {
nhlog : : ui ( ) - > debug ( " WebRTC: screen share frame rate: {} fps " ,
settings - > screenShareFrameRate ( ) ) ;
nhlog : : ui ( ) - > debug ( " WebRTC: screen share picture-in-picture: {} " ,
settings - > screenSharePiP ( ) ) ;
nhlog : : ui ( ) - > debug ( " WebRTC: screen share request remote camera: {} " ,
settings - > screenShareRemoteVideo ( ) ) ;
nhlog : : ui ( ) - > debug ( " WebRTC: screen share hide mouse cursor: {} " ,
settings - > screenShareHideCursor ( ) ) ;
GstElement * ximagesrc = gst_element_factory_make ( " ximagesrc " , " screenshare " ) ;
if ( ! ximagesrc ) {
nhlog : : ui ( ) - > error ( " WebRTC: failed to create ximagesrc " ) ;
return false ;
}
g_object_set ( ximagesrc , " use-damage " , FALSE , nullptr ) ;
g_object_set ( ximagesrc , " xid " , shareWindowId_ , nullptr ) ;
g_object_set (
ximagesrc , " show-pointer " , ! settings - > screenShareHideCursor ( ) , nullptr ) ;
GstCaps * caps = gst_caps_new_simple ( " video/x-raw " ,
" framerate " ,
GST_TYPE_FRACTION ,
settings - > screenShareFrameRate ( ) ,
1 ,
nullptr ) ;
GstElement * capsfilter = gst_element_factory_make ( " capsfilter " , nullptr ) ;
g_object_set ( capsfilter , " caps " , caps , nullptr ) ;
gst_caps_unref ( caps ) ;
gst_bin_add_many ( GST_BIN ( pipe_ ) , ximagesrc , capsfilter , nullptr ) ;
if ( settings - > screenSharePiP ( ) & & devices_ . haveCamera ( ) ) {
GstElement * compositor = gst_element_factory_make ( " compositor " , nullptr ) ;
g_object_set ( compositor , " background " , 1 , nullptr ) ;
gst_bin_add ( GST_BIN ( pipe_ ) , compositor ) ;
if ( ! gst_element_link_many (
ximagesrc , compositor , capsfilter , tee , nullptr ) ) {
nhlog : : ui ( ) - > error ( " WebRTC: failed to link screen share elements " ) ;
return false ;
}
GstPad * srcpad = gst_element_get_static_pad ( camerafilter , " src " ) ;
remotePiPSinkPad_ = gst_element_get_request_pad ( compositor , " sink_%u " ) ;
if ( GST_PAD_LINK_FAILED ( gst_pad_link ( srcpad , remotePiPSinkPad_ ) ) ) {
nhlog : : ui ( ) - > error (
" WebRTC: failed to link camerafilter -> compositor " ) ;
gst_object_unref ( srcpad ) ;
return false ;
}
gst_object_unref ( srcpad ) ;
} else if ( ! gst_element_link_many (
ximagesrc , videoconvert , capsfilter , tee , nullptr ) ) {
nhlog : : ui ( ) - > error ( " WebRTC: failed to link screen share elements " ) ;
return false ;
}
}
GstElement * queue = gst_element_factory_make ( " queue " , nullptr ) ;
GstElement * vp8enc = gst_element_factory_make ( " vp8enc " , nullptr ) ;
g_object_set ( vp8enc , " deadline " , 1 , nullptr ) ;
@ -910,46 +1007,45 @@ WebRTCSession::addVideoPipeline(int vp8PayloadType)
g_object_set ( rtpcapsfilter , " caps " , rtpcaps , nullptr ) ;
gst_caps_unref ( rtpcaps ) ;
gst_bin_add_many ( GST_BIN ( pipe_ ) ,
source ,
videoconvert ,
capsfilter ,
tee ,
queue ,
vp8enc ,
rtpvp8pay ,
rtpqueue ,
rtpcapsfilter ,
nullptr ) ;
gst_bin_add_many (
GST_BIN ( pipe_ ) , queue , vp8enc , rtpvp8pay , rtpqueue , rtpcapsfilter , nullptr ) ;
GstElement * webrtcbin = gst_bin_get_by_name ( GST_BIN ( pipe_ ) , " webrtcbin " ) ;
if ( ! gst_element_link_many ( source ,
videoconvert ,
capsfilter ,
tee ,
queue ,
vp8enc ,
rtpvp8pay ,
rtpqueue ,
rtpcapsfilter ,
webrtcbin ,
nullptr ) ) {
nhlog : : ui ( ) - > error ( " WebRTC: failed to link video pipeline elements " ) ;
if ( ! gst_element_link_many (
tee , queue , vp8enc , rtpvp8pay , rtpqueue , rtpcapsfilter , webrtcbin , nullptr ) ) {
nhlog : : ui ( ) - > error ( " WebRTC: failed to link rtp video elements " ) ;
gst_object_unref ( webrtcbin ) ;
return false ;
}
if ( callType_ = = CallType : : SCREEN & &
! ChatPage : : instance ( ) - > userSettings ( ) - > screenShareRemoteVideo ( ) ) {
GArray * transceivers ;
g_signal_emit_by_name ( webrtcbin , " get-transceivers " , & transceivers ) ;
GstWebRTCRTPTransceiver * transceiver =
g_array_index ( transceivers , GstWebRTCRTPTransceiver * , 1 ) ;
transceiver - > direction = GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY ;
g_array_unref ( transceivers ) ;
}
gst_object_unref ( webrtcbin ) ;
return true ;
}
bool
WebRTCSession : : haveLocalVideo ( ) const
WebRTCSession : : haveLocalPiP ( ) const
{
if ( isVideo_ & & state_ > = State : : INITIATED ) {
GstElement * tee = gst_bin_get_by_name ( GST_BIN ( pipe_ ) , " videosrctee " ) ;
if ( tee ) {
gst_object_unref ( tee ) ;
if ( state_ > = State : : INITIATED ) {
if ( callType_ = = CallType : : VOICE | | isRemoteVideoRecvOnly_ )
return false ;
else if ( callType_ = = CallType : : SCREEN )
return true ;
else {
GstElement * tee = gst_bin_get_by_name ( GST_BIN ( pipe_ ) , " videosrctee " ) ;
if ( tee ) {
gst_object_unref ( tee ) ;
return true ;
}
}
}
return false ;
@ -983,15 +1079,35 @@ WebRTCSession::toggleMicMute()
}
void
WebRTCSession : : toggleCameraView ( )
WebRTCSession : : toggleLocalPiP ( )
{
if ( inset SinkPad_) {
if ( localPiP SinkPad_) {
guint zorder ;
g_object_get ( inset SinkPad_, " zorder " , & zorder , nullptr ) ;
g_object_set ( inset SinkPad_, " zorder " , zorder ? 0 : 2 , nullptr ) ;
g_object_get ( localPiP SinkPad_, " zorder " , & zorder , nullptr ) ;
g_object_set ( localPiP SinkPad_, " zorder " , zorder ? 0 : 2 , nullptr ) ;
}
}
void
WebRTCSession : : clear ( )
{
callType_ = webrtc : : CallType : : VOICE ;
isOffering_ = false ;
isRemoteVideoRecvOnly_ = false ;
isRemoteVideoSendOnly_ = false ;
videoItem_ = nullptr ;
pipe_ = nullptr ;
webrtc_ = nullptr ;
busWatchId_ = 0 ;
shareWindowId_ = 0 ;
haveAudioStream_ = false ;
haveVideoStream_ = false ;
localPiPSinkPad_ = nullptr ;
remotePiPSinkPad_ = nullptr ;
localsdp_ . clear ( ) ;
localcandidates_ . clear ( ) ;
}
void
WebRTCSession : : end ( )
{
@ -1007,12 +1123,7 @@ WebRTCSession::end()
}
}
webrtc_ = nullptr ;
isVideo_ = false ;
isOffering_ = false ;
isRemoteVideoRecvOnly_ = false ;
videoItem_ = nullptr ;
insetSinkPad_ = nullptr ;
clear ( ) ;
if ( state_ ! = State : : DISCONNECTED )
emit stateChanged ( State : : DISCONNECTED ) ;
}
@ -1026,16 +1137,12 @@ WebRTCSession::havePlugins(bool, std::string *)
}
bool
WebRTCSession : : haveLocalVideo ( ) const
WebRTCSession : : haveLocalPiP ( ) const
{
return false ;
}
bool
WebRTCSession : : createOffer ( bool )
{
return false ;
}
bool WebRTCSession : : createOffer ( webrtc : : CallType , uint32_t ) { return false ; }
bool
WebRTCSession : : acceptOffer ( const std : : string & )
@ -1066,7 +1173,7 @@ WebRTCSession::toggleMicMute()
}
void
WebRTCSession : : toggleCameraView ( )
WebRTCSession : : toggleLocalPiP ( )
{ }
void
DeepBlueV7.X
4 years ago
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