Diff
Modified: trunk/LayoutTests/ChangeLog (237909 => 237910)
--- trunk/LayoutTests/ChangeLog 2018-11-07 03:12:18 UTC (rev 237909)
+++ trunk/LayoutTests/ChangeLog 2018-11-07 03:23:59 UTC (rev 237910)
@@ -1,5 +1,15 @@
2018-11-06 Youenn Fablet <you...@apple.com>
+ Calling sender.replaceTrack() twice produces a new transceiver and its corresponding m= section
+ https://bugs.webkit.org/show_bug.cgi?id=191261
+
+ Reviewed by Eric Carlson.
+
+ * webrtc/video-addTransceiver-expected.txt:
+ * webrtc/video-addTransceiver.html:
+
+2018-11-06 Youenn Fablet <you...@apple.com>
+
Make mDNS ICE Candidate an experimental flag again
https://bugs.webkit.org/show_bug.cgi?id=191262
Modified: trunk/LayoutTests/webrtc/video-addTransceiver-expected.txt (237909 => 237910)
--- trunk/LayoutTests/webrtc/video-addTransceiver-expected.txt 2018-11-07 03:12:18 UTC (rev 237909)
+++ trunk/LayoutTests/webrtc/video-addTransceiver-expected.txt 2018-11-07 03:23:59 UTC (rev 237910)
@@ -2,6 +2,7 @@
PASS Setting up calls with addTransceiver but with no track
PASS Setting up calls with addTransceiver with a track
+PASS Setting up calls with addTransceiver with a track and use replaceTrack several times
PASS Basic video exchange set up with addTransceiver
PASS Testing synchronization sources
Modified: trunk/LayoutTests/webrtc/video-addTransceiver.html (237909 => 237910)
--- trunk/LayoutTests/webrtc/video-addTransceiver.html 2018-11-07 03:12:18 UTC (rev 237909)
+++ trunk/LayoutTests/webrtc/video-addTransceiver.html 2018-11-07 03:23:59 UTC (rev 237910)
@@ -39,6 +39,24 @@
assert_true(offer.sdp.indexOf("a=recvonly") !== -1, "a=recvonly");
}, "Setting up calls with addTransceiver with a track");
+promise_test(async (test) => {
+ if (window.testRunner)
+ testRunner.setUserMediaPermission(true);
+
+ const stream = await navigator.mediaDevices.getUserMedia({ video: true });
+ var pc = new RTCPeerConnection();
+ pc.addTransceiver(stream.getVideoTracks()[0]);
+ await pc.getSenders()[0].replaceTrack(null);
+ await pc.getSenders()[0].replaceTrack(stream.getVideoTracks()[0]);
+
+ const offer = await pc.createOffer();
+
+ const sdpLines = offer.sdp.split('\r\n').filter(line => {
+ return line.startsWith("m=video");
+ });
+ assert_equals(sdpLines.length, 1, "There should be 1 video m section");
+}, "Setting up calls with addTransceiver with a track and use replaceTrack several times");
+
function testImage()
{
canvas.width = video.videoWidth;
@@ -79,6 +97,13 @@
secondConnection._ontrack_ = (trackEvent) => {
resolve(trackEvent.track);
};
+ }, {
+ observeOffer : (desc) => {
+ const sdpLines = desc.sdp.split('\r\n').filter(line => {
+ return line.startsWith("m=video");
+ });
+ assert_equals(sdpLines.length, 1, "There should be 1 video m section");
+ }
});
setTimeout(() => reject("Test timed out"), 5000);
});
Modified: trunk/Source/WebCore/ChangeLog (237909 => 237910)
--- trunk/Source/WebCore/ChangeLog 2018-11-07 03:12:18 UTC (rev 237909)
+++ trunk/Source/WebCore/ChangeLog 2018-11-07 03:23:59 UTC (rev 237910)
@@ -1,3 +1,26 @@
+2018-11-06 Youenn Fablet <you...@apple.com>
+
+ Calling sender.replaceTrack() twice produces a new transceiver and its corresponding m= section
+ https://bugs.webkit.org/show_bug.cgi?id=191261
+
+ Reviewed by Eric Carlson.
+
+ Handle the case of replacing a track in a sender that has no track.
+ In particular, do not create a new m-section as was implied by plan B implementation.
+ Instead, set the track directly on the rtc sender.
+ Covered by webrtc/video-addTransceiver.html.
+
+ * Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp:
+ (WebCore::LibWebRTCMediaEndpoint::createSourceAndRTCTrack):
+ (WebCore::LibWebRTCMediaEndpoint::addTransceiver):
+ (WebCore::LibWebRTCMediaEndpoint::setSenderSourceFromTrack):
+ * Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h:
+ * Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp:
+ (WebCore::LibWebRTCPeerConnectionBackend::setSenderSourceFromTrack):
+ * Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h:
+ * Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.cpp:
+ (WebCore::LibWebRTCRtpSenderBackend::replaceTrack):
+
2018-11-06 Chris Dumez <cdu...@apple.com>
Post too much text to iFrame could crash webkit
Modified: trunk/Source/WebCore/Modules/mediastream/RTCRtpSender.cpp (237909 => 237910)
--- trunk/Source/WebCore/Modules/mediastream/RTCRtpSender.cpp 2018-11-07 03:12:18 UTC (rev 237909)
+++ trunk/Source/WebCore/Modules/mediastream/RTCRtpSender.cpp 2018-11-07 03:23:59 UTC (rev 237910)
@@ -92,6 +92,7 @@
return;
}
+ // FIXME: This whole function should be executed as part of the RTCPeerConnection operation queue.
m_backend->replaceTrack(context, *this, WTFMove(withTrack), WTFMove(promise));
}
Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp (237909 => 237910)
--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp 2018-11-07 03:12:18 UTC (rev 237909)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp 2018-11-07 03:23:59 UTC (rev 237910)
@@ -519,13 +519,14 @@
return createTransceiverBackends(type, init, nullptr);
}
-std::optional<LibWebRTCMediaEndpoint::Backends> LibWebRTCMediaEndpoint::addTransceiver(MediaStreamTrack& track, const RTCRtpTransceiverInit& init)
+std::pair<LibWebRTCRtpSenderBackend::Source, rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>> LibWebRTCMediaEndpoint::createSourceAndRTCTrack(MediaStreamTrack& track)
{
LibWebRTCRtpSenderBackend::Source source;
rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> rtcTrack;
switch (track.privateTrack().type()) {
case RealtimeMediaSource::Type::None:
- return std::nullopt;
+ ASSERT_NOT_REACHED();
+ break;
case RealtimeMediaSource::Type::Audio: {
auto audioSource = RealtimeOutgoingAudioSource::create(track.privateTrack());
rtcTrack = m_peerConnectionFactory.CreateAudioTrack(track.id().utf8().data(), audioSource.ptr());
@@ -539,10 +540,22 @@
break;
}
}
+ return std::make_pair(WTFMove(source), WTFMove(rtcTrack));
+}
- return createTransceiverBackends(WTFMove(rtcTrack), init, WTFMove(source));
+std::optional<LibWebRTCMediaEndpoint::Backends> LibWebRTCMediaEndpoint::addTransceiver(MediaStreamTrack& track, const RTCRtpTransceiverInit& init)
+{
+ auto sourceAndTrack = createSourceAndRTCTrack(track);
+ return createTransceiverBackends(WTFMove(sourceAndTrack.second), init, WTFMove(sourceAndTrack.first));
}
+void LibWebRTCMediaEndpoint::setSenderSourceFromTrack(LibWebRTCRtpSenderBackend& sender, MediaStreamTrack& track)
+{
+ auto sourceAndTrack = createSourceAndRTCTrack(track);
+ sender.setSource(WTFMove(sourceAndTrack.first));
+ sender.rtcSender()->SetTrack(WTFMove(sourceAndTrack.second));
+}
+
std::unique_ptr<LibWebRTCRtpTransceiverBackend> LibWebRTCMediaEndpoint::transceiverBackendFromSender(LibWebRTCRtpSenderBackend& backend)
{
for (auto& transceiver : m_backend->GetTransceivers()) {
Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h (237909 => 237910)
--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h 2018-11-07 03:12:18 UTC (rev 237909)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h 2018-11-07 03:23:59 UTC (rev 237910)
@@ -110,6 +110,8 @@
std::optional<Backends> addTransceiver(MediaStreamTrack&, const RTCRtpTransceiverInit&);
std::unique_ptr<LibWebRTCRtpTransceiverBackend> transceiverBackendFromSender(LibWebRTCRtpSenderBackend&);
+ void setSenderSourceFromTrack(LibWebRTCRtpSenderBackend&, MediaStreamTrack&);
+
private:
LibWebRTCMediaEndpoint(LibWebRTCPeerConnectionBackend&, LibWebRTCProvider&);
@@ -163,6 +165,8 @@
: rtc::RefCountReleaseStatus::kDroppedLastRef;
}
+ std::pair<LibWebRTCRtpSenderBackend::Source, rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>> createSourceAndRTCTrack(MediaStreamTrack&);
+
#if !RELEASE_LOG_DISABLED
const Logger& logger() const final { return m_logger.get(); }
const void* logIdentifier() const final { return m_logIdentifier; }
Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp (237909 => 237910)
--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp 2018-11-07 03:12:18 UTC (rev 237909)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp 2018-11-07 03:23:59 UTC (rev 237910)
@@ -471,6 +471,11 @@
return completeAddTransceiver(WTFMove(sender), init, track->id(), track->kind());
}
+void LibWebRTCPeerConnectionBackend::setSenderSourceFromTrack(LibWebRTCRtpSenderBackend& sender, MediaStreamTrack& track)
+{
+ m_endpoint->setSenderSourceFromTrack(sender, track);
+}
+
static inline LibWebRTCRtpTransceiverBackend& backendFromRTPTransceiver(RTCRtpTransceiver& transceiver)
{
return static_cast<LibWebRTCRtpTransceiverBackend&>(*transceiver.backend());
Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h (237909 => 237910)
--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h 2018-11-07 03:12:18 UTC (rev 237909)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h 2018-11-07 03:23:59 UTC (rev 237910)
@@ -37,6 +37,7 @@
class LibWebRTCMediaEndpoint;
class LibWebRTCProvider;
+class LibWebRTCRtpSenderBackend;
class LibWebRTCRtpTransceiverBackend;
class RTCRtpReceiver;
class RTCRtpReceiverBackend;
@@ -93,6 +94,7 @@
ExceptionOr<Ref<RTCRtpTransceiver>> addTransceiver(const String&, const RTCRtpTransceiverInit&) final;
ExceptionOr<Ref<RTCRtpTransceiver>> addTransceiver(Ref<MediaStreamTrack>&&, const RTCRtpTransceiverInit&) final;
+ void setSenderSourceFromTrack(LibWebRTCRtpSenderBackend&, MediaStreamTrack&);
RTCRtpTransceiver* existingTransceiver(WTF::Function<bool(LibWebRTCRtpTransceiverBackend&)>&&);
RTCRtpTransceiver& newRemoteTransceiver(std::unique_ptr<LibWebRTCRtpTransceiverBackend>&&, Ref<RealtimeMediaSource>&&);
Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.cpp (237909 => 237910)
--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.cpp 2018-11-07 03:12:18 UTC (rev 237909)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.cpp 2018-11-07 03:23:59 UTC (rev 237910)
@@ -31,6 +31,7 @@
#include "LibWebRTCUtils.h"
#include "RTCPeerConnection.h"
#include "RTCRtpSender.h"
+#include "RuntimeEnabledFeatures.h"
#include "ScriptExecutionContext.h"
namespace WebCore {
@@ -76,6 +77,7 @@
}
}
+ // FIXME: Remove this postTask once this whole function is executed as part of the RTCPeerConnection operation queue.
context.postTask([protectedSender = makeRef(sender), promise = WTFMove(promise), track = WTFMove(track), this](ScriptExecutionContext&) mutable {
if (protectedSender->isStopped())
return;
@@ -88,14 +90,24 @@
bool hasTrack = protectedSender->track();
protectedSender->setTrack(track.releaseNonNull());
- if (!hasTrack) {
- // FIXME: In case of unified plan, we should use m_rtcSender->SetTrack and no longer need m_peerConnectionBackend.
- auto result = m_peerConnectionBackend->addTrack(*protectedSender->track(), { });
- if (result.hasException()) {
- promise.reject(result.releaseException());
- return;
- }
+
+ if (hasTrack) {
+ promise.resolve();
+ return;
}
+
+ if (RuntimeEnabledFeatures::sharedFeatures().webRTCUnifiedPlanEnabled()) {
+ m_source = nullptr;
+ m_peerConnectionBackend->setSenderSourceFromTrack(*this, *protectedSender->track());
+ promise.resolve();
+ return;
+ }
+
+ auto result = m_peerConnectionBackend->addTrack(*protectedSender->track(), { });
+ if (result.hasException()) {
+ promise.reject(result.releaseException());
+ return;
+ }
promise.resolve();
});
}
