diff --git a/src/WebRTCSession.cpp b/src/WebRTCSession.cpp
index b6d98058..880a14a2 100644
--- a/src/WebRTCSession.cpp
+++ b/src/WebRTCSession.cpp
@@ -1,3 +1,7 @@
+// SPDX-FileCopyrightText: 2021 Nheko Contributors
+//
+// SPDX-License-Identifier: GPL-3.0-or-later
+
#include <QQmlEngine>
#include <QQuickItem>
#include <algorithm>
@@ -10,6 +14,7 @@
#include <thread>
#include <utility>
+#include "CallDevices.h"
#include "ChatPage.h"
#include "Logging.h"
#include "UserSettingsPage.h"
@@ -29,14 +34,20 @@ 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 +93,10 @@ 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 +178,6 @@ 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 +197,6 @@ 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 +204,7 @@ 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 +293,6 @@ 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 +301,6 @@ 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 +328,7 @@ 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(
@@ -382,45 +354,99 @@ 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)
{
GstPad *sinkpad = gst_element_get_static_pad(decodebin, "sink");
@@ -455,7 +481,7 @@ 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 +493,7 @@ 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 +503,9 @@ 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 +524,7 @@ 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 +550,17 @@ 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);
@@ -544,7 +574,6 @@ getMediaAttributes(const GstSDPMessage *sdp,
}
return false;
}
-
}
bool
@@ -603,17 +632,12 @@ 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 +654,15 @@ 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 +675,18 @@ 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 +717,14 @@ 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 +795,10 @@ 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 +880,115 @@ 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 +1010,45 @@ 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;
@@ -983,16 +1082,36 @@ WebRTCSession::toggleMicMute()
}
void
-WebRTCSession::toggleCameraView()
+WebRTCSession::toggleLocalPiP()
{
- if (insetSinkPad_) {
+ if (localPiPSinkPad_) {
guint zorder;
- g_object_get(insetSinkPad_, "zorder", &zorder, nullptr);
- g_object_set(insetSinkPad_, "zorder", zorder ? 0 : 2, nullptr);
+ g_object_get(localPiPSinkPad_, "zorder", &zorder, nullptr);
+ g_object_set(localPiPSinkPad_, "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()
{
nhlog::ui()->debug("WebRTC: ending session");
@@ -1007,12 +1126,7 @@ 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 +1140,12 @@ 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 +1176,7 @@ WebRTCSession::toggleMicMute()
}
void
-WebRTCSession::toggleCameraView()
+WebRTCSession::toggleLocalPiP()
{}
void
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