Diff
Modified: trunk/LayoutTests/ChangeLog (235808 => 235809)
--- trunk/LayoutTests/ChangeLog 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/LayoutTests/ChangeLog 2018-09-07 21:57:47 UTC (rev 235809)
@@ -1,3 +1,32 @@
+2018-09-07 Youenn Fablet <[email protected]>
+
+ Add support for unified plan transceivers
+ https://bugs.webkit.org/show_bug.cgi?id=189390
+
+ Reviewed by Eric Carlson.
+
+ Update tests using mock to enforce plan B for now until unified plan mocks are supported.
+
+ * fast/mediastream/RTCPeerConnection-addIceCandidate-expected.txt:
+ * fast/mediastream/RTCPeerConnection-icecandidate-event.html:
+ * fast/mediastream/RTCPeerConnection-iceconnectionstatechange-event.html:
+ * fast/mediastream/RTCPeerConnection-inspect-answer.html:
+ * fast/mediastream/RTCPeerConnection-inspect-offer.html:
+ * fast/mediastream/RTCPeerConnection-media-setup-single-dialog.html:
+ * fast/mediastream/RTCPeerConnection-media-setup-two-dialogs.html:
+ * fast/mediastream/RTCPeerConnection-remotely-assigned-transceiver-mid.html:
+ * fast/mediastream/RTCPeerConnection-setLocalDescription-offer.html:
+ * fast/mediastream/RTCPeerConnection-setRemoteDescription-offer.html:
+ * webrtc/calling-peerconnection-once-closed.html:
+ Once closed, addTransceiver is expected to throw.
+ * webrtc/libwebrtc/release-while-creating-offer.html:
+ * webrtc/libwebrtc/release-while-getting-stats.html:
+ * webrtc/libwebrtc/release-while-setting-local-description.html:
+ * webrtc/video-getParameters.html: sender and receiver parameters are no longer the same.
+ Testing them individually.
+ * webrtc/video-with-receiver.html:
+ Test is working only with legacy mode.
+
2018-09-07 Frederic Wang <[email protected]>
[CSSOM View] Handle the scrollingElement in Element::scroll(Left/Top/Width/Height/To)
Modified: trunk/LayoutTests/fast/mediastream/RTCPeerConnection-addIceCandidate-expected.txt (235808 => 235809)
--- trunk/LayoutTests/fast/mediastream/RTCPeerConnection-addIceCandidate-expected.txt 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/LayoutTests/fast/mediastream/RTCPeerConnection-addIceCandidate-expected.txt 2018-09-07 21:57:47 UTC (rev 235809)
@@ -11,7 +11,7 @@
*** Define sdpMid, badSdpMid, sdpMLineIndex and badSdpMLineIndex for testing
PASS sdpMLineIndex is not badSdpMLineIndex
-FAIL sdpMid should not be null.
+PASS sdpMid is not null
PASS sdpMid is not badSdpMid
FAIL promise pc.addIceCandidate(new RTCIceCandidate({candidate: validCandidate, sdpMid: badSdpMid})) fulfilled unexpectedly.
FAIL promise pc.addIceCandidate(new RTCIceCandidate({candidate: validCandidate, sdpMLineIndex: badSdpMLineIndex})) fulfilled unexpectedly.
Modified: trunk/LayoutTests/fast/mediastream/RTCPeerConnection-icecandidate-event.html (235808 => 235809)
--- trunk/LayoutTests/fast/mediastream/RTCPeerConnection-icecandidate-event.html 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/LayoutTests/fast/mediastream/RTCPeerConnection-icecandidate-event.html 2018-09-07 21:57:47 UTC (rev 235809)
@@ -8,6 +8,9 @@
<script>
description("Test RTCPeerConnection 'icecandidate' event and gathering done");
+ if (window.testRunner)
+ testRunner.setWebRTCUnifiedPlanEnabled(false);
+
if (window.internals)
internals.useMockRTCPeerConnectionFactory("ICECandidates");
Modified: trunk/LayoutTests/fast/mediastream/RTCPeerConnection-iceconnectionstatechange-event.html (235808 => 235809)
--- trunk/LayoutTests/fast/mediastream/RTCPeerConnection-iceconnectionstatechange-event.html 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/LayoutTests/fast/mediastream/RTCPeerConnection-iceconnectionstatechange-event.html 2018-09-07 21:57:47 UTC (rev 235809)
@@ -7,6 +7,8 @@
<body>
<script>
description("Test RTCPeerConnection 'iceconnectionstatechange' event");
+ if (window.testRunner)
+ testRunner.setWebRTCUnifiedPlanEnabled(false);
if (window.internals)
internals.useMockRTCPeerConnectionFactory("ICEConnectionState");
Modified: trunk/LayoutTests/fast/mediastream/RTCPeerConnection-inspect-answer.html (235808 => 235809)
--- trunk/LayoutTests/fast/mediastream/RTCPeerConnection-inspect-answer.html 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/LayoutTests/fast/mediastream/RTCPeerConnection-inspect-answer.html 2018-09-07 21:57:47 UTC (rev 235809)
@@ -6,6 +6,9 @@
</head>
<body>
<script>
+ if (window.testRunner)
+ testRunner.setWebRTCUnifiedPlanEnabled(false);
+
if (window.internals)
internals.useMockRTCPeerConnectionFactory("");
Modified: trunk/LayoutTests/fast/mediastream/RTCPeerConnection-inspect-offer.html (235808 => 235809)
--- trunk/LayoutTests/fast/mediastream/RTCPeerConnection-inspect-offer.html 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/LayoutTests/fast/mediastream/RTCPeerConnection-inspect-offer.html 2018-09-07 21:57:47 UTC (rev 235809)
@@ -6,6 +6,9 @@
</head>
<body>
<script>
+ if (window.testRunner)
+ testRunner.setWebRTCUnifiedPlanEnabled(false);
+
if (window.internals)
internals.useMockRTCPeerConnectionFactory("");
Modified: trunk/LayoutTests/fast/mediastream/RTCPeerConnection-media-setup-single-dialog.html (235808 => 235809)
--- trunk/LayoutTests/fast/mediastream/RTCPeerConnection-media-setup-single-dialog.html 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/LayoutTests/fast/mediastream/RTCPeerConnection-media-setup-single-dialog.html 2018-09-07 21:57:47 UTC (rev 235809)
@@ -6,6 +6,9 @@
</head>
<body>
<script>
+ if (window.testRunner)
+ testRunner.setWebRTCUnifiedPlanEnabled(false);
+
if (window.internals)
internals.useMockRTCPeerConnectionFactory("");
Modified: trunk/LayoutTests/fast/mediastream/RTCPeerConnection-media-setup-two-dialogs.html (235808 => 235809)
--- trunk/LayoutTests/fast/mediastream/RTCPeerConnection-media-setup-two-dialogs.html 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/LayoutTests/fast/mediastream/RTCPeerConnection-media-setup-two-dialogs.html 2018-09-07 21:57:47 UTC (rev 235809)
@@ -11,9 +11,10 @@
description("Test setting up media between two RTCPeerConnection instances with a single SDP dialog.");
- if (window.testRunner)
+ if (window.testRunner) {
+ testRunner.setWebRTCUnifiedPlanEnabled(false);
testRunner.setUserMediaPermission(true);
- else {
+ } else {
debug("This test can not be run without the testRunner");
finishJSTest();
}
Modified: trunk/LayoutTests/fast/mediastream/RTCPeerConnection-remotely-assigned-transceiver-mid.html (235808 => 235809)
--- trunk/LayoutTests/fast/mediastream/RTCPeerConnection-remotely-assigned-transceiver-mid.html 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/LayoutTests/fast/mediastream/RTCPeerConnection-remotely-assigned-transceiver-mid.html 2018-09-07 21:57:47 UTC (rev 235809)
@@ -6,6 +6,9 @@
</head>
<body>
<script>
+ if (window.testRunner)
+ testRunner.setWebRTCUnifiedPlanEnabled(false);
+
if (window.internals)
internals.useMockRTCPeerConnectionFactory("");
Modified: trunk/LayoutTests/fast/mediastream/RTCPeerConnection-setLocalDescription-offer.html (235808 => 235809)
--- trunk/LayoutTests/fast/mediastream/RTCPeerConnection-setLocalDescription-offer.html 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/LayoutTests/fast/mediastream/RTCPeerConnection-setLocalDescription-offer.html 2018-09-07 21:57:47 UTC (rev 235809)
@@ -6,6 +6,9 @@
</head>
<body>
<script>
+ if (window.testRunner)
+ testRunner.setWebRTCUnifiedPlanEnabled(false);
+
if (window.internals)
internals.useMockRTCPeerConnectionFactory("");
Modified: trunk/LayoutTests/fast/mediastream/RTCPeerConnection-setRemoteDescription-offer.html (235808 => 235809)
--- trunk/LayoutTests/fast/mediastream/RTCPeerConnection-setRemoteDescription-offer.html 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/LayoutTests/fast/mediastream/RTCPeerConnection-setRemoteDescription-offer.html 2018-09-07 21:57:47 UTC (rev 235809)
@@ -6,6 +6,9 @@
</head>
<body>
<script>
+ if (window.testRunner)
+ testRunner.setWebRTCUnifiedPlanEnabled(false);
+
if (window.internals)
internals.useMockRTCPeerConnectionFactory("");
Modified: trunk/LayoutTests/imported/w3c/ChangeLog (235808 => 235809)
--- trunk/LayoutTests/imported/w3c/ChangeLog 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/LayoutTests/imported/w3c/ChangeLog 2018-09-07 21:57:47 UTC (rev 235809)
@@ -1,3 +1,19 @@
+2018-09-07 Youenn Fablet <[email protected]>
+
+ Add support for unified plan transceivers
+ https://bugs.webkit.org/show_bug.cgi?id=189390
+
+ Reviewed by Eric Carlson.
+
+ * web-platform-tests/webrtc/RTCDTMFSender-insertDTMF.https-expected.txt:
+ * web-platform-tests/webrtc/RTCPeerConnection-addTrack.https-expected.txt:
+ Regression comes from the fact that the sender was never used but transceiver direction is sendrecv.
+ This might need further clarification in the spec or implementation of libwebrtc.
+ * web-platform-tests/webrtc/RTCPeerConnection-addTransceiver-expected.txt:
+ * web-platform-tests/webrtc/RTCPeerConnection-removeTrack.https-expected.txt:
+ * web-platform-tests/webrtc/RTCPeerConnection-setDescription-transceiver-expected.txt:
+ * web-platform-tests/webrtc/RTCRtpTransceiver-setDirection-expected.txt:
+
2018-09-07 Rob Buis <[email protected]>
XMLHttpRequest: open() does not throw a SYNTAX_ERR exception if method is empty or url cannot be resolved
Modified: trunk/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCDTMFSender-insertDTMF.https-expected.txt (235808 => 235809)
--- trunk/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCDTMFSender-insertDTMF.https-expected.txt 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCDTMFSender-insertDTMF.https-expected.txt 2018-09-07 21:57:47 UTC (rev 235809)
@@ -2,8 +2,8 @@
FAIL insertDTMF() should succeed if tones contains valid DTMF characters promise_test: Unhandled rejection with value: object "ReferenceError: Can't find variable: RTCDTMFSender"
FAIL insertDTMF() should throw InvalidCharacterError if tones contains invalid DTMF characters promise_test: Unhandled rejection with value: object "ReferenceError: Can't find variable: RTCDTMFSender"
FAIL insertDTMF() should throw InvalidStateError if transceiver is stopped assert_throws: function "() => dtmfSender.insertDTMF('')" threw object "TypeError: undefined is not an object (evaluating 'dtmfSender.insertDTMF')" that is not a DOMException InvalidStateError: property "code" is equal to undefined, expected 11
-FAIL insertDTMF() should throw InvalidStateError if transceiver.currentDirection is recvonly assert_equals: expected (string) "inactive" but got (undefined) undefined
-FAIL insertDTMF() should throw InvalidStateError if transceiver.currentDirection is inactive assert_equals: expected (string) "inactive" but got (undefined) undefined
+FAIL insertDTMF() should throw InvalidStateError if transceiver.currentDirection is recvonly assert_equals: expected (string) "inactive" but got (object) null
+FAIL insertDTMF() should throw InvalidStateError if transceiver.currentDirection is inactive assert_equals: expected (string) "inactive" but got (object) null
FAIL insertDTMF() should set toneBuffer to provided tones normalized, with old tones overridden promise_test: Unhandled rejection with value: object "ReferenceError: Can't find variable: RTCDTMFSender"
FAIL insertDTMF() after remove and close should reject assert_throws: function "() =>
dtmfSender.insertDTMF('123')" threw object "TypeError: undefined is not an object (evaluating 'dtmfSender.insertDTMF')" that is not a DOMException InvalidStateError: property "code" is equal to undefined, expected 11
Modified: trunk/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-addTrack.https-expected.txt (235808 => 235809)
--- trunk/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-addTrack.https-expected.txt 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-addTrack.https-expected.txt 2018-09-07 21:57:47 UTC (rev 235809)
@@ -5,6 +5,6 @@
PASS addTrack with single track argument and multiple mediaStreams should succeed
PASS Adding the same track multiple times should throw InvalidAccessError
PASS addTrack with existing sender with null track, same kind, and recvonly direction should reuse sender
-PASS addTrack with existing sender with null track, same kind, and sendrecv direction should create new sender
+FAIL addTrack with existing sender with null track, same kind, and sendrecv direction should create new sender assert_not_equals: got disallowed value object "[object RTCRtpSender]"
PASS addTrack with existing sender with null track, different kind, and recvonly direction should create new sender
Modified: trunk/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-addTransceiver-expected.txt (235808 => 235809)
--- trunk/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-addTransceiver-expected.txt 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-addTransceiver-expected.txt 2018-09-07 21:57:47 UTC (rev 235809)
@@ -1,11 +1,11 @@
PASS addTransceiver() with string argument as invalid kind should throw TypeError
-FAIL addTransceiver('audio') should return an audio transceiver assert_equals: expected (object) null but got (undefined) undefined
-FAIL addTransceiver('video') should return a video transceiver assert_equals: expected "remote video" but got ""
+PASS addTransceiver('audio') should return an audio transceiver
+PASS addTransceiver('video') should return a video transceiver
PASS addTransceiver() with direction sendonly should have result transceiver.direction be the same
PASS addTransceiver() with direction inactive should have result transceiver.direction be the same
PASS addTransceiver() with invalid direction should throw TypeError
-FAIL addTransceiver(track) should have result with sender.track be given track assert_equals: expected "remote audio" but got ""
+PASS addTransceiver(track) should have result with sender.track be given track
PASS addTransceiver(track) multiple times should create multiple transceivers
FAIL addTransceiver() with rid containing invalid non-alphanumeric characters should throw TypeError assert_throws: function "() =>
pc.addTransceiver('audio', {
Modified: trunk/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-onnegotiationneeded-expected.txt (235808 => 235809)
--- trunk/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-onnegotiationneeded-expected.txt 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-onnegotiationneeded-expected.txt 2018-09-07 21:57:47 UTC (rev 235809)
@@ -3,9 +3,9 @@
PASS Creating first data channel should fire negotiationneeded event
PASS calling createDataChannel twice should fire negotiationneeded event once
-TIMEOUT addTransceiver() should fire negotiationneeded event Test timed out
-PASS Calling addTransceiver() twice should fire negotiationneeded event once
-PASS Calling both addTransceiver() and createDataChannel() should fire negotiationneeded event once
+PASS addTransceiver() should fire negotiationneeded event
+FAIL Calling addTransceiver() twice should fire negotiationneeded event once assert_unreached: Pending promise should never be resolved. Instead it is fulfilled with: [object Object] Reached unreachable code
+FAIL Calling both addTransceiver() and createDataChannel() should fire negotiationneeded event once assert_unreached: Pending promise should never be resolved. Instead it is fulfilled with: [object Object] Reached unreachable code
PASS negotiationneeded event should not fire if signaling state is not stable
-NOTRUN negotiationneeded event should fire only after signaling state go back to stable
+TIMEOUT negotiationneeded event should fire only after signaling state go back to stable Test timed out
Modified: trunk/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-removeTrack.https-expected.txt (235808 => 235809)
--- trunk/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-removeTrack.https-expected.txt 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-removeTrack.https-expected.txt 2018-09-07 21:57:47 UTC (rev 235809)
@@ -5,10 +5,10 @@
PASS addTrack - Calling removeTrack on different connection that is closed should throw InvalidStateError
FAIL addTransceiver - Calling removeTrack on different connection should throw InvalidAccessError assert_throws: function "() => pc2.removeTrack(sender)" did not throw
FAIL addTrack - Calling removeTrack on different connection should throw InvalidAccessError assert_throws: function "() => pc2.removeTrack(sender)" did not throw
-FAIL addTransceiver - Calling removeTrack with valid sender should set sender.track to null assert_equals: expected (object) null but got (undefined) undefined
+FAIL addTransceiver - Calling removeTrack with valid sender should set sender.track to null assert_equals: direction should not be altered expected "sendrecv" but got "recvonly"
PASS addTrack - Calling removeTrack with valid sender should set sender.track to null
-FAIL Calling removeTrack with currentDirection sendrecv should set direction to recvonly assert_equals: expected (object) null but got (undefined) undefined
-FAIL Calling removeTrack with currentDirection sendonly should set direction to inactive assert_equals: expected (object) null but got (undefined) undefined
-FAIL Calling removeTrack with currentDirection recvonly should not change direction assert_equals: expected (object) null but got (undefined) undefined
-FAIL Calling removeTrack with currentDirection inactive should not change direction assert_equals: expected (object) null but got (undefined) undefined
+FAIL Calling removeTrack with currentDirection sendrecv should set direction to recvonly assert_equals: expected "sendrecv" but got "sendonly"
+PASS Calling removeTrack with currentDirection sendonly should set direction to inactive
+FAIL Calling removeTrack with currentDirection recvonly should not change direction assert_equals: expected "recvonly" but got "inactive"
+PASS Calling removeTrack with currentDirection inactive should not change direction
Modified: trunk/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-setDescription-transceiver-expected.txt (235808 => 235809)
--- trunk/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-setDescription-transceiver-expected.txt 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-setDescription-transceiver-expected.txt 2018-09-07 21:57:47 UTC (rev 235809)
@@ -1,7 +1,7 @@
-FAIL setLocalDescription(offer) with m= section should assign mid to corresponding transceiver assert_equals: Expect transceiver.mid to set to valid string value expected "string" but got "object"
+PASS setLocalDescription(offer) with m= section should assign mid to corresponding transceiver
FAIL setRemoteDescription(offer) with m= section and no existing transceiver should create corresponding transceiver promise_test: Unhandled rejection with value: object "TypeError: pc2.setRemoteDescrption is not a function. (In 'pc2.setRemoteDescrption(offer)', 'pc2.setRemoteDescrption' is undefined)"
-FAIL setLocalDescription(rollback) should unset transceiver.mid assert_not_equals: got disallowed value null
-FAIL setLocalDescription(rollback) should only unset transceiver mids associated with current round assert_not_equals: got disallowed value null
-FAIL setRemoteDescription(rollback) should remove newly created transceiver from transceiver list assert_equals: expected 1 but got 0
+FAIL setLocalDescription(rollback) should unset transceiver.mid promise_test: Unhandled rejection with value: object "InvalidStateError: Description type incompatible with current signaling state"
+FAIL setLocalDescription(rollback) should only unset transceiver mids associated with current round promise_test: Unhandled rejection with value: object "InvalidStateError: Description type incompatible with current signaling state"
+FAIL setRemoteDescription(rollback) should remove newly created transceiver from transceiver list promise_test: Unhandled rejection with value: object "InvalidStateError: Description type incompatible with current signaling state"
Modified: trunk/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpTransceiver-setDirection-expected.txt (235808 => 235809)
--- trunk/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpTransceiver-setDirection-expected.txt 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpTransceiver-setDirection-expected.txt 2018-09-07 21:57:47 UTC (rev 235809)
@@ -1,5 +1,5 @@
-FAIL setDirection should change transceiver.direction assert_equals: expected (object) null but got (undefined) undefined
+PASS setDirection should change transceiver.direction
PASS setDirection with same direction should have no effect
-FAIL setDirection should change transceiver.direction independent of transceiver.currentDirection assert_equals: expected (object) null but got (undefined) undefined
+FAIL setDirection should change transceiver.direction independent of transceiver.currentDirection assert_equals: expected "recvonly" but got "inactive"
Modified: trunk/LayoutTests/webrtc/calling-peerconnection-once-closed.html (235808 => 235809)
--- trunk/LayoutTests/webrtc/calling-peerconnection-once-closed.html 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/LayoutTests/webrtc/calling-peerconnection-once-closed.html 2018-09-07 21:57:47 UTC (rev 235809)
@@ -42,7 +42,7 @@
promise_test(() => {
return closedConnection().then((connection) => {
- connection.addTransceiver("video");
+ assert_throws("InvalidStateError", () => { connection.addTransceiver("video"); });
});
}, "Ensuring closed connection addTransceiver does not crash");
Modified: trunk/LayoutTests/webrtc/libwebrtc/release-while-creating-offer.html (235808 => 235809)
--- trunk/LayoutTests/webrtc/libwebrtc/release-while-creating-offer.html 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/LayoutTests/webrtc/libwebrtc/release-while-creating-offer.html 2018-09-07 21:57:47 UTC (rev 235809)
@@ -7,6 +7,9 @@
<script>
self.jsTestIsAsync = true;
+if (window.testRunner)
+ testRunner.setWebRTCUnifiedPlanEnabled(false);
+
if (window.internals)
internals.useMockRTCPeerConnectionFactory("LibWebRTCReleasingWhileCreatingOffer");
Modified: trunk/LayoutTests/webrtc/libwebrtc/release-while-getting-stats.html (235808 => 235809)
--- trunk/LayoutTests/webrtc/libwebrtc/release-while-getting-stats.html 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/LayoutTests/webrtc/libwebrtc/release-while-getting-stats.html 2018-09-07 21:57:47 UTC (rev 235809)
@@ -7,6 +7,9 @@
<script>
self.jsTestIsAsync = true;
+if (window.testRunner)
+ testRunner.setWebRTCUnifiedPlanEnabled(false);
+
if (window.internals)
internals.useMockRTCPeerConnectionFactory("LibWebRTCReleasingWhileGettingStats");
Modified: trunk/LayoutTests/webrtc/libwebrtc/release-while-setting-local-description.html (235808 => 235809)
--- trunk/LayoutTests/webrtc/libwebrtc/release-while-setting-local-description.html 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/LayoutTests/webrtc/libwebrtc/release-while-setting-local-description.html 2018-09-07 21:57:47 UTC (rev 235809)
@@ -10,6 +10,9 @@
// Silence unhandled rejection messages.
window._onunhandledrejection_ = () => false;
+if (window.testRunner)
+ testRunner.setWebRTCUnifiedPlanEnabled(false);
+
if (window.internals)
internals.useMockRTCPeerConnectionFactory("LibWebRTCReleasingWhileSettingDescription");
Modified: trunk/LayoutTests/webrtc/video-getParameters.html (235808 => 235809)
--- trunk/LayoutTests/webrtc/video-getParameters.html 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/LayoutTests/webrtc/video-getParameters.html 2018-09-07 21:57:47 UTC (rev 235809)
@@ -42,16 +42,13 @@
senderParameters.encodings[0].fec.ssrc = 1;
senderParameters.encodings[0].rtx.ssrc = 1;
- receiverParameters.encodings[0].fec.ssrc = 1;
- receiverParameters.encodings[0].rtx.ssrc = 1;
senderParameters.transactionId = "";
receiverParameters.transactionId = "";
- assert_equals(JSON.stringify(senderParameters), JSON.stringify(receiverParameters), "testing sender vs. receiver parameters");
-
senderParameters.encodings[0].ssrc = 1;
- assert_equals(JSON.stringify(senderParameters), '{"codecs":[],"degradationPreference":"balanced","encodings":[{"active":true,"fec":{"ssrc":1},"maxBitrate":0,"maxFramerate":0,"priority":"medium","rid":"","rtx":{"ssrc":1},"scaleResolutionDownBy":1,"ssrc":1}],"headerExtensions":[],"transactionId":""}', "Testing sanitized parameters");
+ assert_equals(JSON.stringify(senderParameters), '{"codecs":[],"degradationPreference":"balanced","encodings":[{"active":true,"fec":{"ssrc":1},"maxBitrate":0,"maxFramerate":0,"priority":"medium","rid":"","rtx":{"ssrc":1},"scaleResolutionDownBy":1,"ssrc":1}],"headerExtensions":[],"transactionId":""}', "Testing sanitized sender parameters");
+ assert_equals(JSON.stringify(receiverParameters), '{"codecs":[],"degradationPreference":"balanced","encodings":[],"headerExtensions":[],"transactionId":""}', "Testing sanitized receiver parameters");
});
}, "Basic video stats");
</script>
Modified: trunk/LayoutTests/webrtc/video-with-receiver.html (235808 => 235809)
--- trunk/LayoutTests/webrtc/video-with-receiver.html 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/LayoutTests/webrtc/video-with-receiver.html 2018-09-07 21:57:47 UTC (rev 235809)
@@ -14,6 +14,9 @@
video = document.getElementById("video");
canvas = document.getElementById("canvas");
+if (window.testRunner)
+ testRunner.setWebRTCUnifiedPlanEnabled(false);
+
function testImage()
{
canvas.width = video.videoWidth;
Modified: trunk/Source/ThirdParty/libwebrtc/ChangeLog (235808 => 235809)
--- trunk/Source/ThirdParty/libwebrtc/ChangeLog 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/Source/ThirdParty/libwebrtc/ChangeLog 2018-09-07 21:57:47 UTC (rev 235809)
@@ -1,3 +1,15 @@
+2018-09-07 Youenn Fablet <[email protected]>
+
+ Add support for unified plan transceivers
+ https://bugs.webkit.org/show_bug.cgi?id=189390
+
+ Reviewed by Eric Carlson.
+
+ Expose more symbols.
+ * Configurations/libwebrtc.iOS.exp:
+ * Configurations/libwebrtc.iOSsim.exp:
+ * Configurations/libwebrtc.mac.exp:
+
2018-09-05 Youenn Fablet <[email protected]>
Expose RTCRtpSender.setParameters
Modified: trunk/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOS.exp (235808 => 235809)
--- trunk/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOS.exp 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOS.exp 2018-09-07 21:57:47 UTC (rev 235809)
@@ -212,3 +212,5 @@
__ZN6webrtc21RtpEncodingParametersC1ERKS0_
__ZN6webrtc21RtpEncodingParametersC1Ev
__ZN6webrtc21RtpEncodingParametersD1Ev
+__ZN6webrtc18RtpTransceiverInitC1Ev
+__ZN6webrtc18RtpTransceiverInitD1Ev
Modified: trunk/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOSsim.exp (235808 => 235809)
--- trunk/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOSsim.exp 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOSsim.exp 2018-09-07 21:57:47 UTC (rev 235809)
@@ -213,3 +213,6 @@
__ZN6webrtc21RtpEncodingParametersC1ERKS0_
__ZN6webrtc21RtpEncodingParametersC1Ev
__ZN6webrtc21RtpEncodingParametersD1Ev
+
+__ZN6webrtc18RtpTransceiverInitC1Ev
+__ZN6webrtc18RtpTransceiverInitD1Ev
Modified: trunk/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.mac.exp (235808 => 235809)
--- trunk/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.mac.exp 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.mac.exp 2018-09-07 21:57:47 UTC (rev 235809)
@@ -213,3 +213,5 @@
__ZN6webrtc21RtpEncodingParametersC1ERKS0_
__ZN6webrtc21RtpEncodingParametersC1Ev
__ZN6webrtc21RtpEncodingParametersD1Ev
+__ZN6webrtc18RtpTransceiverInitC1Ev
+__ZN6webrtc18RtpTransceiverInitD1Ev
Modified: trunk/Source/WebCore/CMakeLists.txt (235808 => 235809)
--- trunk/Source/WebCore/CMakeLists.txt 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/Source/WebCore/CMakeLists.txt 2018-09-07 21:57:47 UTC (rev 235809)
@@ -1572,6 +1572,7 @@
Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp
Modules/mediastream/libwebrtc/LibWebRTCRtpReceiverBackend.cpp
Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.cpp
+ Modules/mediastream/libwebrtc/LibWebRTCRtpTransceiverBackend.cpp
Modules/mediastream/libwebrtc/LibWebRTCUtils.cpp
)
endif ()
Modified: trunk/Source/WebCore/ChangeLog (235808 => 235809)
--- trunk/Source/WebCore/ChangeLog 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/Source/WebCore/ChangeLog 2018-09-07 21:57:47 UTC (rev 235809)
@@ -1,3 +1,83 @@
+2018-09-07 Youenn Fablet <[email protected]>
+
+ Add support for unified plan transceivers
+ https://bugs.webkit.org/show_bug.cgi?id=189390
+
+ Reviewed by Eric Carlson.
+
+ Keep previous transceiver behavior when unified plan flag is off.
+ Otherwise, use the libwebrtc transceiver API to create and use unified plan transceivers.
+ Fuel the implementation of transceivers through a dedicated backend.
+
+ Update transceiver IDL and make some smaller fixes at the same time:
+ - Make sure remote sources have a proper name as per https://w3c.github.io/webrtc-pc/#dfn-create-an-rtcrtpreceiver.
+ - Add support for transceiver.currentDirection.
+
+ Our mock peer connections are only supporting plan B APIs at the moment.
+ We therefore mandate plan B when using such mocks until we can upgrade mocks to support unified plan APIs.
+
+ Covered by modified and rebased tests.
+
+ * Modules/mediastream/PeerConnectionBackend.cpp:
+ (WebCore::PeerConnectionBackend::addTrack):
+ * Modules/mediastream/PeerConnectionBackend.h:
+ * Modules/mediastream/RTCPeerConnection.cpp:
+ (WebCore::RTCPeerConnection::addTrack):
+ (WebCore::RTCPeerConnection::addTransceiver):
+ * Modules/mediastream/RTCRtpSender.cpp:
+ (WebCore::RTCRtpSender::RTCRtpSender):
+ * Modules/mediastream/RTCRtpTransceiver.cpp:
+ (WebCore::RTCRtpTransceiver::mid const):
+ (WebCore::RTCRtpTransceiver::currentDirection const):
+ * Modules/mediastream/RTCRtpTransceiver.h:
+ (WebCore::RTCRtpTransceiver::backend):
+ * Modules/mediastream/RTCRtpTransceiver.idl:
+ * Modules/mediastream/RTCRtpTransceiverBackend.h:
+ * Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp:
+ (WebCore::LibWebRTCMediaEndpoint::addTrack):
+ (WebCore::LibWebRTCMediaEndpoint::newTransceiver):
+ (WebCore::LibWebRTCMediaEndpoint::addTransceiver):
+ (WebCore::LibWebRTCMediaEndpoint::transceiverBackendFromSender):
+ (WebCore::LibWebRTCMediaEndpoint::OnAddTrack):
+ (WebCore::LibWebRTCMediaEndpoint::OnTrack):
+ * Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h:
+ * Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp:
+ (WebCore::createReceiverForSource):
+ (WebCore::LibWebRTCPeerConnectionBackend::createReceiver):
+ (WebCore::LibWebRTCPeerConnectionBackend::videoReceiver):
+ (WebCore::LibWebRTCPeerConnectionBackend::audioReceiver):
+ (WebCore::LibWebRTCPeerConnectionBackend::addTrack):
+ (WebCore::LibWebRTCPeerConnectionBackend::addTransceiver):
+ (WebCore::backendFromRTPTransceiver):
+ (WebCore::LibWebRTCPeerConnectionBackend::existingTransceiver):
+ (WebCore::LibWebRTCPeerConnectionBackend::newTransceiver):
+ * Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h:
+ * Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.cpp:
+ (WebCore::LibWebRTCRtpSenderBackend::replaceTrack):
+ * Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.h:
+ * Modules/mediastream/libwebrtc/LibWebRTCRtpTransceiverBackend.cpp: Added.
+ (WebCore::LibWebRTCRtpTransceiverBackend::createReceiverBackend):
+ (WebCore::LibWebRTCRtpTransceiverBackend::createSenderBackend):
+ (WebCore::LibWebRTCRtpTransceiverBackend::direction const):
+ (WebCore::LibWebRTCRtpTransceiverBackend::currentDirection const):
+ (WebCore::LibWebRTCRtpTransceiverBackend::setDirection):
+ (WebCore::LibWebRTCRtpTransceiverBackend::mid):
+ (WebCore::LibWebRTCRtpTransceiverBackend::stop):
+ * Modules/mediastream/libwebrtc/LibWebRTCRtpTransceiverBackend.h: Added.
+ * Modules/mediastream/libwebrtc/LibWebRTCUtils.cpp:
+ (WebCore::toRTCRtpTransceiverDirection):
+ (WebCore::fromRTCRtpTransceiverDirection):
+ (WebCore::fromRtpTransceiverInit):
+ * Modules/mediastream/libwebrtc/LibWebRTCUtils.h:
+ * WebCore.xcodeproj/project.pbxproj:
+ * platform/mediastream/RealtimeIncomingAudioSource.cpp:
+ (WebCore::RealtimeIncomingAudioSource::RealtimeIncomingAudioSource):
+ * platform/mediastream/RealtimeIncomingVideoSource.cpp:
+ (WebCore::RealtimeIncomingVideoSource::RealtimeIncomingVideoSource):
+ * platform/mediastream/RealtimeMediaSource.h:
+ * testing/Internals.cpp:
+ (WebCore::Internals::useMockRTCPeerConnectionFactory):
+
2018-09-07 Rob Buis <[email protected]>
XMLHttpRequest: open() does not throw a SYNTAX_ERR exception if method is empty or url cannot be resolved
Modified: trunk/Source/WebCore/Modules/mediastream/PeerConnectionBackend.cpp (235808 => 235809)
--- trunk/Source/WebCore/Modules/mediastream/PeerConnectionBackend.cpp 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/Source/WebCore/Modules/mediastream/PeerConnectionBackend.cpp 2018-09-07 21:57:47 UTC (rev 235809)
@@ -530,7 +530,7 @@
m_peerConnection.scheduleNegotiationNeededEvent();
}
-ExceptionOr<Ref<RTCRtpSender>> PeerConnectionBackend::addTrack(RTCRtpSender*, MediaStreamTrack&, const Vector<String>&)
+ExceptionOr<Ref<RTCRtpSender>> PeerConnectionBackend::addTrack(MediaStreamTrack&, Vector<String>&&)
{
return Exception { NotSupportedError, "Not implemented"_s };
}
Modified: trunk/Source/WebCore/Modules/mediastream/PeerConnectionBackend.h (235808 => 235809)
--- trunk/Source/WebCore/Modules/mediastream/PeerConnectionBackend.h 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/Source/WebCore/Modules/mediastream/PeerConnectionBackend.h 2018-09-07 21:57:47 UTC (rev 235809)
@@ -100,7 +100,7 @@
virtual void getStats(MediaStreamTrack*, Ref<DeferredPromise>&&) = 0;
- virtual ExceptionOr<Ref<RTCRtpSender>> addTrack(RTCRtpSender*, MediaStreamTrack&, const Vector<String>&);
+ virtual ExceptionOr<Ref<RTCRtpSender>> addTrack(MediaStreamTrack&, Vector<String>&&);
virtual void removeTrack(RTCRtpSender&) { }
virtual ExceptionOr<Ref<RTCRtpTransceiver>> addTransceiver(const String&, const RTCRtpTransceiverInit&);
Modified: trunk/Source/WebCore/Modules/mediastream/RTCPeerConnection.cpp (235808 => 235809)
--- trunk/Source/WebCore/Modules/mediastream/RTCPeerConnection.cpp 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/Source/WebCore/Modules/mediastream/RTCPeerConnection.cpp 2018-09-07 21:57:47 UTC (rev 235809)
@@ -122,22 +122,7 @@
for (auto stream : streams)
mediaStreamIds.append(stream.get().id());
- RTCRtpSender* sender = nullptr;
-
- // Reuse an existing sender with the same track kind if it has never been used to send before.
- for (auto& transceiver : m_transceiverSet->list()) {
- auto& existingSender = transceiver->sender();
- if (existingSender.trackKind() == track->kind() && existingSender.trackId().isNull() && !transceiver->hasSendingDirection()) {
- existingSender.setTrack(track.copyRef());
- existingSender.setMediaStreamIds(WTFMove(mediaStreamIds));
- transceiver->enableSendingDirection();
- sender = &existingSender;
-
- break;
- }
- }
-
- return m_backend->addTrack(sender, track.get(), mediaStreamIds);
+ return m_backend->addTrack(track.get(), WTFMove(mediaStreamIds));
}
ExceptionOr<void> RTCPeerConnection::removeTrack(RTCRtpSender& sender)
@@ -168,12 +153,18 @@
if (WTF::holds_alternative<String>(withTrack)) {
const String& kind = WTF::get<String>(withTrack);
- if (kind != "audio" && kind != "video")
+ if (kind != "audio"_s && kind != "video"_s)
return Exception { TypeError };
+ if (isClosed())
+ return Exception { InvalidStateError };
+
return m_backend->addTransceiver(kind, init);
}
+ if (isClosed())
+ return Exception { InvalidStateError };
+
auto track = WTF::get<RefPtr<MediaStreamTrack>>(withTrack).releaseNonNull();
return m_backend->addTransceiver(WTFMove(track), init);
}
Modified: trunk/Source/WebCore/Modules/mediastream/RTCRtpReceiver.h (235808 => 235809)
--- trunk/Source/WebCore/Modules/mediastream/RTCRtpReceiver.h 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/Source/WebCore/Modules/mediastream/RTCRtpReceiver.h 2018-09-07 21:57:47 UTC (rev 235809)
@@ -41,7 +41,7 @@
class RTCRtpReceiver : public RTCRtpSenderReceiverBase {
public:
- static Ref<RTCRtpReceiver> create(Ref<MediaStreamTrack>&& track, std::unique_ptr<RTCRtpReceiverBackend>&& backend = nullptr)
+ static Ref<RTCRtpReceiver> create(Ref<MediaStreamTrack>&& track, std::unique_ptr<RTCRtpReceiverBackend>&& backend)
{
return adoptRef(*new RTCRtpReceiver(WTFMove(track), WTFMove(backend)));
}
Modified: trunk/Source/WebCore/Modules/mediastream/RTCRtpSender.cpp (235808 => 235809)
--- trunk/Source/WebCore/Modules/mediastream/RTCRtpSender.cpp 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/Source/WebCore/Modules/mediastream/RTCRtpSender.cpp 2018-09-07 21:57:47 UTC (rev 235809)
@@ -33,6 +33,8 @@
#if ENABLE(WEB_RTC)
+#include "RuntimeEnabledFeatures.h"
+
namespace WebCore {
Ref<RTCRtpSender> RTCRtpSender::create(Ref<MediaStreamTrack>&& track, Vector<String>&& mediaStreamIds, std::unique_ptr<RTCRtpSenderBackend>&& backend)
@@ -53,6 +55,7 @@
, m_mediaStreamIds(WTFMove(mediaStreamIds))
, m_backend(WTFMove(backend))
{
+ ASSERT(!RuntimeEnabledFeatures::sharedFeatures().webRTCUnifiedPlanEnabled() || m_backend);
}
void RTCRtpSender::setTrackToNull()
Modified: trunk/Source/WebCore/Modules/mediastream/RTCRtpTransceiver.cpp (235808 => 235809)
--- trunk/Source/WebCore/Modules/mediastream/RTCRtpTransceiver.cpp 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/Source/WebCore/Modules/mediastream/RTCRtpTransceiver.cpp 2018-09-07 21:57:47 UTC (rev 235809)
@@ -47,11 +47,9 @@
{
}
-const String& RTCRtpTransceiver::mid() const
+String RTCRtpTransceiver::mid() const
{
- if (!m_backend)
- return m_mid;
- return m_backend->mid();
+ return m_backend ? m_backend->mid() : String { };
}
bool RTCRtpTransceiver::hasSendingDirection() const
@@ -66,6 +64,13 @@
return m_backend->direction();
}
+std::optional<RTCRtpTransceiverDirection> RTCRtpTransceiver::currentDirection() const
+{
+ if (!m_backend)
+ return std::nullopt;
+ return m_backend->currentDirection();
+}
+
void RTCRtpTransceiver::setDirection(RTCRtpTransceiverDirection direction)
{
m_direction = direction;
Modified: trunk/Source/WebCore/Modules/mediastream/RTCRtpTransceiver.h (235808 => 235809)
--- trunk/Source/WebCore/Modules/mediastream/RTCRtpTransceiver.h 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/Source/WebCore/Modules/mediastream/RTCRtpTransceiver.h 2018-09-07 21:57:47 UTC (rev 235809)
@@ -55,8 +55,9 @@
void disableSendingDirection();
RTCRtpTransceiverDirection direction() const;
+ std::optional<RTCRtpTransceiverDirection> currentDirection() const;
void setDirection(RTCRtpTransceiverDirection);
- const String& mid() const;
+ String mid() const;
RTCRtpSender& sender() { return m_sender.get(); }
RTCRtpReceiver& receiver() { return m_receiver.get(); }
@@ -69,10 +70,11 @@
// transport each.
RTCIceTransport& iceTransport() { return m_iceTransport.get(); }
+ RTCRtpTransceiverBackend* backend() { return m_backend.get(); }
+
private:
RTCRtpTransceiver(Ref<RTCRtpSender>&&, Ref<RTCRtpReceiver>&&, std::unique_ptr<RTCRtpTransceiverBackend>&&);
- String m_mid;
RTCRtpTransceiverDirection m_direction;
Ref<RTCRtpSender> m_sender;
Modified: trunk/Source/WebCore/Modules/mediastream/RTCRtpTransceiver.idl (235808 => 235809)
--- trunk/Source/WebCore/Modules/mediastream/RTCRtpTransceiver.idl 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/Source/WebCore/Modules/mediastream/RTCRtpTransceiver.idl 2018-09-07 21:57:47 UTC (rev 235809)
@@ -36,13 +36,11 @@
EnabledAtRuntime=PeerConnection
] interface RTCRtpTransceiver {
readonly attribute DOMString? mid;
- // FIXME 169662: missing [SameObject]
- readonly attribute RTCRtpSender sender;
- // FIXME 169662: missing [SameObject]
- readonly attribute RTCRtpReceiver receiver;
+ [SameObject] readonly attribute RTCRtpSender sender;
+ [SameObject] readonly attribute RTCRtpReceiver receiver;
readonly attribute boolean stopped;
readonly attribute RtpTransceiverDirection direction;
- // FIXME 169662: missing currentDirection
+ readonly attribute RTCRtpTransceiverDirection? currentDirection;
void setDirection(RtpTransceiverDirection direction);
void stop();
// FIXME 169662: missing setCodecPreferences
Modified: trunk/Source/WebCore/Modules/mediastream/RTCRtpTransceiverBackend.h (235808 => 235809)
--- trunk/Source/WebCore/Modules/mediastream/RTCRtpTransceiverBackend.h 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/Source/WebCore/Modules/mediastream/RTCRtpTransceiverBackend.h 2018-09-07 21:57:47 UTC (rev 235809)
@@ -35,9 +35,10 @@
virtual ~RTCRtpTransceiverBackend() = default;
virtual RTCRtpTransceiverDirection direction() const = 0;
+ virtual std::optional<RTCRtpTransceiverDirection> currentDirection() const = 0;
virtual void setDirection(RTCRtpTransceiverDirection) = 0;
- virtual const String& mid() = 0;
+ virtual String mid() = 0;
virtual void stop() = 0;
};
Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp (235808 => 235809)
--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp 2018-09-07 21:57:47 UTC (rev 235809)
@@ -34,6 +34,7 @@
#include "LibWebRTCProvider.h"
#include "LibWebRTCRtpReceiverBackend.h"
#include "LibWebRTCRtpSenderBackend.h"
+#include "LibWebRTCRtpTransceiverBackend.h"
#include "LibWebRTCStatsCollector.h"
#include "LibWebRTCUtils.h"
#include "Logging.h"
@@ -197,7 +198,6 @@
bool LibWebRTCMediaEndpoint::addTrack(LibWebRTCRtpSenderBackend& sender, MediaStreamTrack& track, const Vector<String>& mediaStreamIds)
{
ASSERT(m_backend);
- ASSERT(!sender.rtcSender());
if (!RuntimeEnabledFeatures::sharedFeatures().webRTCUnifiedPlanEnabled()) {
String mediaStreamId = mediaStreamIds.isEmpty() ? createCanonicalUUIDString() : mediaStreamIds[0];
@@ -208,35 +208,41 @@
});
}
- std::vector<std::string> ids;
- for (auto& id : mediaStreamIds)
- ids.push_back(id.utf8().data());
-
+ LibWebRTCRtpSenderBackend::Source source;
+ rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> rtcTrack;
switch (track.privateTrack().type()) {
case RealtimeMediaSource::Type::Audio: {
auto audioSource = RealtimeOutgoingAudioSource::create(track.privateTrack());
- auto audioTrack = m_peerConnectionFactory.CreateAudioTrack(track.id().utf8().data(), audioSource.ptr());
- sender.setSource(WTFMove(audioSource));
- auto rtpSender = m_backend->AddTrack(audioTrack.get(), WTFMove(ids));
- if (!rtpSender.ok())
- return false;
- sender.setRTCSender(rtpSender.MoveValue());
- return true;
+ rtcTrack = m_peerConnectionFactory.CreateAudioTrack(track.id().utf8().data(), audioSource.ptr());
+ source = WTFMove(audioSource);
+ break;
}
case RealtimeMediaSource::Type::Video: {
auto videoSource = RealtimeOutgoingVideoSource::create(track.privateTrack());
- auto videoTrack = m_peerConnectionFactory.CreateVideoTrack(track.id().utf8().data(), videoSource.ptr());
- sender.setSource(WTFMove(videoSource));
- auto rtpSender = m_backend->AddTrack(videoTrack.get(), WTFMove(ids));
- if (!rtpSender.ok())
- return false;
- sender.setRTCSender(rtpSender.MoveValue());
- return true;
+ rtcTrack = m_peerConnectionFactory.CreateVideoTrack(track.id().utf8().data(), videoSource.ptr());
+ source = WTFMove(videoSource);
+ break;
}
case RealtimeMediaSource::Type::None:
ASSERT_NOT_REACHED();
+ return false;
}
- return false;
+
+ sender.setSource(WTFMove(source));
+ if (auto rtpSender = sender.rtcSender()) {
+ rtpSender->SetTrack(rtcTrack.get());
+ return true;
+ }
+
+ std::vector<std::string> ids;
+ for (auto& id : mediaStreamIds)
+ ids.push_back(id.utf8().data());
+
+ auto newRTPSender = m_backend->AddTrack(rtcTrack.get(), WTFMove(ids));
+ if (!newRTPSender.ok())
+ return false;
+ sender.setRTCSender(newRTPSender.MoveValue());
+ return true;
}
void LibWebRTCMediaEndpoint::removeTrack(LibWebRTCRtpSenderBackend& sender)
@@ -361,20 +367,131 @@
}
receiver->setBackend(std::make_unique<LibWebRTCRtpReceiverBackend>(WTFMove(rtcReceiver)));
-
- auto* track = receiver->track();
- ASSERT(track);
+ auto& track = *receiver->track();
+ fireTrackEvent(receiver.releaseNonNull(), track, rtcStreams, nullptr);
+}
+void LibWebRTCMediaEndpoint::fireTrackEvent(Ref<RTCRtpReceiver>&& receiver, Ref<MediaStreamTrack>&& track, const std::vector<rtc::scoped_refptr<webrtc::MediaStreamInterface>>& rtcStreams, RefPtr<RTCRtpTransceiver>&& transceiver)
+{
Vector<RefPtr<MediaStream>> streams;
for (auto& rtcStream : rtcStreams) {
auto& mediaStream = mediaStreamFromRTCStream(*rtcStream.get());
streams.append(&mediaStream);
- mediaStream.addTrackFromPlatform(*track);
+ mediaStream.addTrackFromPlatform(track.get());
}
m_peerConnectionBackend.connection().fireEvent(RTCTrackEvent::create(eventNames().trackEvent,
- Event::CanBubble::No, Event::IsCancelable::No, WTFMove(receiver), track, WTFMove(streams), nullptr));
+ Event::CanBubble::No, Event::IsCancelable::No, WTFMove(receiver), WTFMove(track), WTFMove(streams), WTFMove(transceiver)));
}
+static inline void setExistingReceiverSourceTrack(RealtimeMediaSource& existingSource, webrtc::RtpReceiverInterface& rtcReceiver)
+{
+ switch (rtcReceiver.media_type()) {
+ case cricket::MEDIA_TYPE_AUDIO: {
+ ASSERT(existingSource.type() == RealtimeMediaSource::Type::Audio);
+ rtc::scoped_refptr<webrtc::AudioTrackInterface> audioTrack = static_cast<webrtc::AudioTrackInterface*>(rtcReceiver.track().get());
+ static_cast<RealtimeIncomingAudioSource&>(existingSource).setSourceTrack(WTFMove(audioTrack));
+ return;
+ }
+ case cricket::MEDIA_TYPE_VIDEO: {
+ ASSERT(existingSource.type() == RealtimeMediaSource::Type::Video);
+ rtc::scoped_refptr<webrtc::VideoTrackInterface> videoTrack = static_cast<webrtc::VideoTrackInterface*>(rtcReceiver.track().get());
+ static_cast<RealtimeIncomingVideoSource&>(existingSource).setSourceTrack(WTFMove(videoTrack));
+ return;
+ }
+ case cricket::MEDIA_TYPE_DATA:
+ ASSERT_NOT_REACHED();
+ return;
+ }
+}
+
+static inline RefPtr<RealtimeMediaSource> sourceFromNewReceiver(webrtc::RtpReceiverInterface& rtcReceiver)
+{
+ auto rtcTrack = rtcReceiver.track();
+ switch (rtcReceiver.media_type()) {
+ case cricket::MEDIA_TYPE_DATA:
+ return nullptr;
+ case cricket::MEDIA_TYPE_AUDIO: {
+ rtc::scoped_refptr<webrtc::AudioTrackInterface> audioTrack = static_cast<webrtc::AudioTrackInterface*>(rtcTrack.get());
+ return RealtimeIncomingAudioSource::create(WTFMove(audioTrack), fromStdString(rtcTrack->id()));
+ }
+ case cricket::MEDIA_TYPE_VIDEO: {
+ rtc::scoped_refptr<webrtc::VideoTrackInterface> videoTrack = static_cast<webrtc::VideoTrackInterface*>(rtcTrack.get());
+ return RealtimeIncomingVideoSource::create(WTFMove(videoTrack), fromStdString(rtcTrack->id()));
+ }
+ }
+}
+
+void LibWebRTCMediaEndpoint::newTransceiver(rtc::scoped_refptr<webrtc::RtpTransceiverInterface>&& rtcTransceiver)
+{
+ auto* transceiver = m_peerConnectionBackend.existingTransceiver([&](auto& transceiverBackend) {
+ return rtcTransceiver.get() == transceiverBackend.rtcTransceiver();
+ });
+ if (transceiver) {
+ setExistingReceiverSourceTrack(transceiver->receiver().track()->source(), *rtcTransceiver->receiver());
+ return;
+ }
+
+ auto rtcReceiver = rtcTransceiver->receiver();
+ auto source = sourceFromNewReceiver(*rtcReceiver);
+ if (!source)
+ return;
+
+ auto& newTransceiver = m_peerConnectionBackend.newRemoteTransceiver(std::make_unique<LibWebRTCRtpTransceiverBackend>(WTFMove(rtcTransceiver)), source.releaseNonNull());
+
+ fireTrackEvent(makeRef(newTransceiver.receiver()), *newTransceiver.receiver().track(), rtcReceiver->streams(), makeRef(newTransceiver));
+}
+
+template<typename T>
+std::optional<LibWebRTCMediaEndpoint::Backends> LibWebRTCMediaEndpoint::createTransceiverBackends(T&& trackOrKind, const RTCRtpTransceiverInit& init, LibWebRTCRtpSenderBackend::Source&& source)
+{
+ auto result = m_backend->AddTransceiver(WTFMove(trackOrKind), fromRtpTransceiverInit(init));
+ if (!result.ok())
+ return std::nullopt;
+
+ auto transceiver = std::make_unique<LibWebRTCRtpTransceiverBackend>(result.MoveValue());
+ return LibWebRTCMediaEndpoint::Backends { transceiver->createSenderBackend(m_peerConnectionBackend, WTFMove(source)), transceiver->createReceiverBackend(), WTFMove(transceiver) };
+}
+
+std::optional<LibWebRTCMediaEndpoint::Backends> LibWebRTCMediaEndpoint::addTransceiver(const String& trackKind, const RTCRtpTransceiverInit& init)
+{
+ auto type = trackKind == "audio" ? cricket::MediaType::MEDIA_TYPE_AUDIO : cricket::MediaType::MEDIA_TYPE_VIDEO;
+ return createTransceiverBackends(type, init, nullptr);
+}
+
+std::optional<LibWebRTCMediaEndpoint::Backends> LibWebRTCMediaEndpoint::addTransceiver(MediaStreamTrack& track, const RTCRtpTransceiverInit& init)
+{
+ LibWebRTCRtpSenderBackend::Source source;
+ rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> rtcTrack;
+ switch (track.privateTrack().type()) {
+ case RealtimeMediaSource::Type::None:
+ return std::nullopt;
+ case RealtimeMediaSource::Type::Audio: {
+ auto audioSource = RealtimeOutgoingAudioSource::create(track.privateTrack());
+ rtcTrack = m_peerConnectionFactory.CreateAudioTrack(track.id().utf8().data(), audioSource.ptr());
+ source = WTFMove(audioSource);
+ break;
+ }
+ case RealtimeMediaSource::Type::Video: {
+ auto videoSource = RealtimeOutgoingVideoSource::create(track.privateTrack());
+ rtcTrack = m_peerConnectionFactory.CreateVideoTrack(track.id().utf8().data(), videoSource.ptr());
+ source = WTFMove(videoSource);
+ break;
+ }
+ }
+
+ return createTransceiverBackends(WTFMove(rtcTrack), init, WTFMove(source));
+}
+
+std::unique_ptr<LibWebRTCRtpTransceiverBackend> LibWebRTCMediaEndpoint::transceiverBackendFromSender(LibWebRTCRtpSenderBackend& backend)
+{
+ for (auto& transceiver : m_backend->GetTransceivers()) {
+ if (transceiver->sender().get() == backend.rtcSender())
+ return std::make_unique<LibWebRTCRtpTransceiverBackend>(rtc::scoped_refptr<webrtc::RtpTransceiverInterface>(transceiver));
+ }
+ return nullptr;
+}
+
+
void LibWebRTCMediaEndpoint::removeRemoteStream(webrtc::MediaStreamInterface& rtcStream)
{
bool removed = m_streams.remove(&rtcStream);
@@ -403,6 +520,9 @@
void LibWebRTCMediaEndpoint::OnAddTrack(rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver, const std::vector<rtc::scoped_refptr<webrtc::MediaStreamInterface>>& streams)
{
+ if (RuntimeEnabledFeatures::sharedFeatures().webRTCUnifiedPlanEnabled())
+ return;
+
callOnMainThread([protectedThis = makeRef(*this), receiver = WTFMove(receiver), streams]() mutable {
if (protectedThis->isStopped())
return;
@@ -410,6 +530,19 @@
});
}
+void LibWebRTCMediaEndpoint::OnTrack(rtc::scoped_refptr<webrtc::RtpTransceiverInterface> transceiver)
+{
+ if (!RuntimeEnabledFeatures::sharedFeatures().webRTCUnifiedPlanEnabled())
+ return;
+
+ callOnMainThread([protectedThis = makeRef(*this), transceiver = WTFMove(transceiver)]() mutable {
+ if (protectedThis->isStopped())
+ return;
+ protectedThis->newTransceiver(WTFMove(transceiver));
+ });
+}
+
+
std::unique_ptr<RTCDataChannelHandler> LibWebRTCMediaEndpoint::createDataChannel(const String& label, const RTCDataChannelInit& options)
{
auto init = LibWebRTCDataChannelHandler::fromRTCDataChannelInit(options);
Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h (235808 => 235809)
--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h 2018-09-07 21:57:47 UTC (rev 235809)
@@ -28,10 +28,8 @@
#include "LibWebRTCObservers.h"
#include "LibWebRTCProvider.h"
-#include "PeerConnectionBackend.h"
+#include "LibWebRTCRtpSenderBackend.h"
#include "RTCRtpReceiver.h"
-#include "RealtimeOutgoingAudioSource.h"
-#include "RealtimeOutgoingVideoSource.h"
#include <Timer.h>
#pragma GCC diagnostic push
@@ -61,7 +59,8 @@
class LibWebRTCProvider;
class LibWebRTCPeerConnectionBackend;
-class LibWebRTCRtpSenderBackend;
+class LibWebRTCRtpReceiverBackend;
+class LibWebRTCRtpTransceiverBackend;
class MediaStreamTrack;
class RTCSessionDescription;
@@ -101,6 +100,15 @@
bool addTrack(LibWebRTCRtpSenderBackend&, MediaStreamTrack&, const Vector<String>&);
void removeTrack(LibWebRTCRtpSenderBackend&);
+ struct Backends {
+ std::unique_ptr<LibWebRTCRtpSenderBackend> senderBackend;
+ std::unique_ptr<LibWebRTCRtpReceiverBackend> receiverBackend;
+ std::unique_ptr<LibWebRTCRtpTransceiverBackend> transceiverBackend;
+ };
+ std::optional<Backends> addTransceiver(const String& trackKind, const RTCRtpTransceiverInit&);
+ std::optional<Backends> addTransceiver(MediaStreamTrack&, const RTCRtpTransceiverInit&);
+ std::unique_ptr<LibWebRTCRtpTransceiverBackend> transceiverBackendFromSender(LibWebRTCRtpSenderBackend&);
+
private:
LibWebRTCMediaEndpoint(LibWebRTCPeerConnectionBackend&, LibWebRTCProvider&);
@@ -110,6 +118,8 @@
void OnRemoveStream(rtc::scoped_refptr<webrtc::MediaStreamInterface>) final;
void OnDataChannel(rtc::scoped_refptr<webrtc::DataChannelInterface>) final;
void OnAddTrack(rtc::scoped_refptr<webrtc::RtpReceiverInterface>, const std::vector<rtc::scoped_refptr<webrtc::MediaStreamInterface>>&) final;
+ void OnTrack(rtc::scoped_refptr<webrtc::RtpTransceiverInterface>) final;
+
void OnRenegotiationNeeded() final;
void OnIceConnectionChange(webrtc::PeerConnectionInterface::IceConnectionState) final;
void OnIceGatheringChange(webrtc::PeerConnectionInterface::IceGatheringState) final;
@@ -125,7 +135,13 @@
void addRemoteStream(webrtc::MediaStreamInterface&);
void addRemoteTrack(rtc::scoped_refptr<webrtc::RtpReceiverInterface>&&, const std::vector<rtc::scoped_refptr<webrtc::MediaStreamInterface>>&);
void removeRemoteStream(webrtc::MediaStreamInterface&);
+ void newTransceiver(rtc::scoped_refptr<webrtc::RtpTransceiverInterface>&&);
+ void fireTrackEvent(Ref<RTCRtpReceiver>&&, Ref<MediaStreamTrack>&&, const std::vector<rtc::scoped_refptr<webrtc::MediaStreamInterface>>&, RefPtr<RTCRtpTransceiver>&&);
+
+ template<typename T>
+ std::optional<Backends> createTransceiverBackends(T&&, const RTCRtpTransceiverInit&, LibWebRTCRtpSenderBackend::Source&&);
+
void OnStatsDelivered(const rtc::scoped_refptr<const webrtc::RTCStatsReport>&) final;
void gatherStatsForLogging();
void startLoggingStats();
Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp (235808 => 235809)
--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp 2018-09-07 21:57:47 UTC (rev 235809)
@@ -33,6 +33,7 @@
#include "LibWebRTCMediaEndpoint.h"
#include "LibWebRTCRtpReceiverBackend.h"
#include "LibWebRTCRtpSenderBackend.h"
+#include "LibWebRTCRtpTransceiverBackend.h"
#include "MediaEndpointConfiguration.h"
#include "Page.h"
#include "RTCIceCandidate.h"
@@ -200,12 +201,12 @@
addIceCandidateSucceeded();
}
-static inline Ref<RTCRtpReceiver> createReceiverForSource(ScriptExecutionContext& context, Ref<RealtimeMediaSource>&& source)
+static inline Ref<RTCRtpReceiver> createReceiverForSource(ScriptExecutionContext& context, Ref<RealtimeMediaSource>&& source, std::unique_ptr<RTCRtpReceiverBackend>&& backend)
{
auto remoteTrackPrivate = MediaStreamTrackPrivate::create(WTFMove(source), String { source->id() });
auto remoteTrack = MediaStreamTrack::create(context, WTFMove(remoteTrackPrivate));
- return RTCRtpReceiver::create(WTFMove(remoteTrack));
+ return RTCRtpReceiver::create(WTFMove(remoteTrack), WTFMove(backend));
}
static inline Ref<RealtimeMediaSource> createEmptySource(const String& trackKind, String&& trackId)
@@ -219,7 +220,7 @@
Ref<RTCRtpReceiver> LibWebRTCPeerConnectionBackend::createReceiver(const String& trackKind, const String& trackId)
{
- auto receiver = createReceiverForSource(*m_peerConnection.scriptExecutionContext(), createEmptySource(trackKind, String(trackId)));
+ auto receiver = createReceiverForSource(*m_peerConnection.scriptExecutionContext(), createEmptySource(trackKind, String(trackId)), nullptr);
m_pendingReceivers.append(receiver.copyRef());
return receiver;
}
@@ -237,7 +238,7 @@
}
}
auto source = RealtimeIncomingVideoSource::create(nullptr, WTFMove(trackId));
- auto receiver = createReceiverForSource(*m_peerConnection.scriptExecutionContext(), source.copyRef());
+ auto receiver = createReceiverForSource(*m_peerConnection.scriptExecutionContext(), source.copyRef(), nullptr);
auto senderBackend = std::make_unique<LibWebRTCRtpSenderBackend>(*this, nullptr);
auto transceiver = RTCRtpTransceiver::create(RTCRtpSender::create("video", { }, WTFMove(senderBackend)), receiver.copyRef(), nullptr);
@@ -260,7 +261,7 @@
}
}
auto source = RealtimeIncomingAudioSource::create(nullptr, WTFMove(trackId));
- auto receiver = createReceiverForSource(*m_peerConnection.scriptExecutionContext(), source.copyRef());
+ auto receiver = createReceiverForSource(*m_peerConnection.scriptExecutionContext(), source.copyRef(), nullptr);
auto senderBackend = std::make_unique<LibWebRTCRtpSenderBackend>(*this, nullptr);
auto transceiver = RTCRtpTransceiver::create(RTCRtpSender::create("audio", { }, WTFMove(senderBackend)), receiver.copyRef(), nullptr);
@@ -316,11 +317,57 @@
static inline LibWebRTCRtpSenderBackend& backendFromRTPSender(RTCRtpSender& sender)
{
+ ASSERT(!sender.isStopped());
return static_cast<LibWebRTCRtpSenderBackend&>(*sender.backend());
}
-ExceptionOr<Ref<RTCRtpSender>> LibWebRTCPeerConnectionBackend::addTrack(RTCRtpSender* sender, MediaStreamTrack& track, const Vector<String>& mediaStreamIds)
+
+static inline RefPtr<RTCRtpSender> findExistingSender(const Vector<std::reference_wrapper<RTCRtpSender>>& senders, LibWebRTCRtpSenderBackend& senderBackend)
{
+ ASSERT(senderBackend.rtcSender());
+ for (RTCRtpSender& sender : senders) {
+ if (!sender.isStopped() && senderBackend.rtcSender() == backendFromRTPSender(sender).rtcSender())
+ return makeRef(sender);
+ }
+ return nullptr;
+}
+
+ExceptionOr<Ref<RTCRtpSender>> LibWebRTCPeerConnectionBackend::addTrack(MediaStreamTrack& track, Vector<String>&& mediaStreamIds)
+{
+ if (RuntimeEnabledFeatures::sharedFeatures().webRTCUnifiedPlanEnabled()) {
+ auto senderBackend = std::make_unique<LibWebRTCRtpSenderBackend>(*this, nullptr);
+ if (!m_endpoint->addTrack(*senderBackend, track, mediaStreamIds))
+ return Exception { TypeError, "Unable to add track"_s };
+
+ if (auto sender = findExistingSender(m_peerConnection.getSenders(), *senderBackend)) {
+ sender->setTrack(makeRef(track));
+ sender->setMediaStreamIds(WTFMove(mediaStreamIds));
+ return sender.releaseNonNull();
+ }
+
+ auto transceiverBackend = m_endpoint->transceiverBackendFromSender(*senderBackend);
+
+ auto sender = RTCRtpSender::create(makeRef(track), WTFMove(mediaStreamIds), WTFMove(senderBackend));
+ auto receiver = createReceiverForSource(*m_peerConnection.scriptExecutionContext(), createEmptySource(track.kind(), createCanonicalUUIDString()), transceiverBackend->createReceiverBackend());
+ auto transceiver = RTCRtpTransceiver::create(sender.copyRef(), WTFMove(receiver), WTFMove(transceiverBackend));
+ m_peerConnection.addInternalTransceiver(WTFMove(transceiver));
+ return WTFMove(sender);
+ }
+
+ RTCRtpSender* sender = nullptr;
+ // Reuse an existing sender with the same track kind if it has never been used to send before.
+ for (auto& transceiver : m_peerConnection.getTransceivers()) {
+ auto& existingSender = transceiver->sender();
+ if (!existingSender.isStopped() && existingSender.trackKind() == track.kind() && existingSender.trackId().isNull() && !transceiver->hasSendingDirection()) {
+ existingSender.setTrack(makeRef(track));
+ existingSender.setMediaStreamIds(WTFMove(mediaStreamIds));
+ transceiver->enableSendingDirection();
+ sender = &existingSender;
+
+ break;
+ }
+ }
+
if (!sender) {
const String& trackKind = track.kind();
String trackId = createCanonicalUUIDString();
@@ -340,8 +387,25 @@
return makeRef(*sender);
}
+template<typename T>
+ExceptionOr<Ref<RTCRtpTransceiver>> LibWebRTCPeerConnectionBackend::addUnifiedPlanTransceiver(T&& trackOrKind, const RTCRtpTransceiverInit& init)
+{
+ auto backends = m_endpoint->addTransceiver(trackOrKind, init);
+ if (!backends)
+ return Exception { InvalidAccessError, "Unable to add transceiver"_s };
+
+ auto sender = RTCRtpSender::create(WTFMove(trackOrKind), Vector<String> { }, WTFMove(backends->senderBackend));
+ auto receiver = createReceiverForSource(*m_peerConnection.scriptExecutionContext(), createEmptySource(sender->trackKind(), createCanonicalUUIDString()), WTFMove(backends->receiverBackend));
+ auto transceiver = RTCRtpTransceiver::create(WTFMove(sender), WTFMove(receiver), WTFMove(backends->transceiverBackend));
+ m_peerConnection.addInternalTransceiver(transceiver.copyRef());
+ return WTFMove(transceiver);
+}
+
ExceptionOr<Ref<RTCRtpTransceiver>> LibWebRTCPeerConnectionBackend::addTransceiver(const String& trackKind, const RTCRtpTransceiverInit& init)
{
+ if (RuntimeEnabledFeatures::sharedFeatures().webRTCUnifiedPlanEnabled())
+ return addUnifiedPlanTransceiver(String { trackKind }, init);
+
auto senderBackend = std::make_unique<LibWebRTCRtpSenderBackend>(*this, nullptr);
auto newSender = RTCRtpSender::create(String(trackKind), Vector<String>(), WTFMove(senderBackend));
return completeAddTransceiver(WTFMove(newSender), init, createCanonicalUUIDString(), trackKind);
@@ -349,6 +413,9 @@
ExceptionOr<Ref<RTCRtpTransceiver>> LibWebRTCPeerConnectionBackend::addTransceiver(Ref<MediaStreamTrack>&& track, const RTCRtpTransceiverInit& init)
{
+ if (RuntimeEnabledFeatures::sharedFeatures().webRTCUnifiedPlanEnabled())
+ return addUnifiedPlanTransceiver(WTFMove(track), init);
+
auto senderBackend = std::make_unique<LibWebRTCRtpSenderBackend>(*this, nullptr);
auto& backend = *senderBackend;
auto sender = RTCRtpSender::create(track.copyRef(), Vector<String>(), WTFMove(senderBackend));
@@ -358,6 +425,29 @@
return completeAddTransceiver(WTFMove(sender), init, track->id(), track->kind());
}
+static inline LibWebRTCRtpTransceiverBackend& backendFromRTPTransceiver(RTCRtpTransceiver& transceiver)
+{
+ return static_cast<LibWebRTCRtpTransceiverBackend&>(*transceiver.backend());
+}
+
+RTCRtpTransceiver* LibWebRTCPeerConnectionBackend::existingTransceiver(WTF::Function<bool(LibWebRTCRtpTransceiverBackend&)>&& matchingFunction)
+{
+ for (auto& transceiver : m_peerConnection.getTransceivers()) {
+ if (matchingFunction(backendFromRTPTransceiver(*transceiver)))
+ return transceiver.get();
+ }
+ return nullptr;
+}
+
+RTCRtpTransceiver& LibWebRTCPeerConnectionBackend::newRemoteTransceiver(std::unique_ptr<LibWebRTCRtpTransceiverBackend>&& transceiverBackend, Ref<RealtimeMediaSource>&& receiverSource)
+{
+ auto sender = RTCRtpSender::create(receiverSource->type() == RealtimeMediaSource::Type::Audio ? "audio"_s : "video"_s, Vector<String> { }, transceiverBackend->createSenderBackend(*this, nullptr));
+ auto receiver = createReceiverForSource(*m_peerConnection.scriptExecutionContext(), WTFMove(receiverSource), transceiverBackend->createReceiverBackend());
+ auto transceiver = RTCRtpTransceiver::create(WTFMove(sender), WTFMove(receiver), WTFMove(transceiverBackend));
+ m_peerConnection.addInternalTransceiver(transceiver.copyRef());
+ return transceiver.get();
+}
+
Ref<RTCRtpTransceiver> LibWebRTCPeerConnectionBackend::completeAddTransceiver(Ref<RTCRtpSender>&& sender, const RTCRtpTransceiverInit& init, const String& trackId, const String& trackKind)
{
auto transceiver = RTCRtpTransceiver::create(WTFMove(sender), createReceiver(trackKind, trackId), nullptr);
@@ -373,21 +463,13 @@
m_endpoint->removeTrack(backendFromRTPSender(sender));
}
-template<typename Source>
-static inline bool updateTrackSource(Source& source, MediaStreamTrack* track)
-{
- if (!track) {
- source.stop();
- return true;
- }
- return source.setSource(track->privateTrack());
-}
-
void LibWebRTCPeerConnectionBackend::applyRotationForOutgoingVideoSources()
{
for (auto& transceiver : m_peerConnection.getTransceivers()) {
- if (auto* videoSource = backendFromRTPSender(transceiver->sender()).videoSource())
- videoSource->setApplyRotation(true);
+ if (!transceiver->sender().isStopped()) {
+ if (auto* videoSource = backendFromRTPSender(transceiver->sender()).videoSource())
+ videoSource->setApplyRotation(true);
+ }
}
}
Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h (235808 => 235809)
--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h 2018-09-07 21:57:47 UTC (rev 235809)
@@ -38,11 +38,13 @@
class LibWebRTCMediaEndpoint;
class LibWebRTCProvider;
+class LibWebRTCRtpTransceiverBackend;
class RTCRtpReceiver;
class RTCSessionDescription;
class RTCStatsReport;
class RealtimeIncomingAudioSource;
class RealtimeIncomingVideoSource;
+class RealtimeMediaSource;
class RealtimeOutgoingAudioSource;
class RealtimeOutgoingVideoSource;
@@ -84,12 +86,15 @@
void getStatsSucceeded(const DeferredPromise&, Ref<RTCStatsReport>&&);
- ExceptionOr<Ref<RTCRtpSender>> addTrack(RTCRtpSender*, MediaStreamTrack&, const Vector<String>&) final;
+ ExceptionOr<Ref<RTCRtpSender>> addTrack(MediaStreamTrack&, Vector<String>&&) final;
void removeTrack(RTCRtpSender&) final;
ExceptionOr<Ref<RTCRtpTransceiver>> addTransceiver(const String&, const RTCRtpTransceiverInit&) final;
ExceptionOr<Ref<RTCRtpTransceiver>> addTransceiver(Ref<MediaStreamTrack>&&, const RTCRtpTransceiverInit&) final;
+ RTCRtpTransceiver* existingTransceiver(WTF::Function<bool(LibWebRTCRtpTransceiverBackend&)>&&);
+ RTCRtpTransceiver& newRemoteTransceiver(std::unique_ptr<LibWebRTCRtpTransceiverBackend>&&, Ref<RealtimeMediaSource>&&);
+
struct VideoReceiver {
Ref<RTCRtpReceiver> receiver;
Ref<RealtimeIncomingVideoSource> source;
@@ -108,6 +113,9 @@
Ref<RTCRtpReceiver> createReceiver(const String& trackKind, const String& trackId);
+ template<typename T>
+ ExceptionOr<Ref<RTCRtpTransceiver>> addUnifiedPlanTransceiver(T&& trackOrKind, const RTCRtpTransceiverInit&);
+
Ref<LibWebRTCMediaEndpoint> m_endpoint;
bool m_isLocalDescriptionSet { false };
bool m_isRemoteDescriptionSet { false };
Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.cpp (235808 => 235809)
--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.cpp 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.cpp 2018-09-07 21:57:47 UTC (rev 235809)
@@ -90,7 +90,7 @@
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.ptr(), *protectedSender->track(), { });
+ auto result = m_peerConnectionBackend->addTrack(*protectedSender->track(), { });
if (result.hasException()) {
promise.reject(result.releaseException());
return;
Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.h (235808 => 235809)
--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.h 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.h 2018-09-07 21:57:47 UTC (rev 235809)
@@ -27,6 +27,7 @@
#if ENABLE(WEB_RTC)
#include "LibWebRTCMacros.h"
+#include "LibWebRTCPeerConnectionBackend.h"
#include "RTCRtpSenderBackend.h"
#include "RealtimeOutgoingAudioSource.h"
#include "RealtimeOutgoingVideoSource.h"
@@ -46,12 +47,20 @@
class LibWebRTCRtpSenderBackend final : public RTCRtpSenderBackend {
public:
- explicit LibWebRTCRtpSenderBackend(LibWebRTCPeerConnectionBackend& backend, rtc::scoped_refptr<webrtc::RtpSenderInterface>&& rtcSender)
+ LibWebRTCRtpSenderBackend(LibWebRTCPeerConnectionBackend& backend, rtc::scoped_refptr<webrtc::RtpSenderInterface>&& rtcSender)
: m_peerConnectionBackend(makeWeakPtr(&backend))
, m_rtcSender(WTFMove(rtcSender))
{
}
+ using Source = Variant<std::nullptr_t, Ref<RealtimeOutgoingAudioSource>, Ref<RealtimeOutgoingVideoSource>>;
+ LibWebRTCRtpSenderBackend(LibWebRTCPeerConnectionBackend& backend, rtc::scoped_refptr<webrtc::RtpSenderInterface>&& rtcSender, Source&& source)
+ : m_peerConnectionBackend(makeWeakPtr(&backend))
+ , m_rtcSender(WTFMove(rtcSender))
+ , m_source(WTFMove(source))
+ {
+ }
+
void setRTCSender(rtc::scoped_refptr<webrtc::RtpSenderInterface>&& rtcSender) { m_rtcSender = WTFMove(rtcSender); }
webrtc::RtpSenderInterface* rtcSender() { return m_rtcSender.get(); }
@@ -71,26 +80,21 @@
);
}
- bool hasNoSource() const
+ bool hasSource() const
{
return WTF::switchOn(m_source,
- [] (const std::nullptr_t&) { return true; },
- [] (const auto&) { return false; }
+ [] (const std::nullptr_t&) { return false; },
+ [] (const auto&) { return true; }
);
}
- void setSource(Ref<RealtimeOutgoingAudioSource>&& source)
+ void setSource(Source&& source)
{
- ASSERT(hasNoSource());
+ ASSERT(!hasSource());
m_source = WTFMove(source);
+ ASSERT(hasSource());
}
- void setSource(Ref<RealtimeOutgoingVideoSource>&& source)
- {
- ASSERT(hasNoSource());
- m_source = WTFMove(source);
- }
-
private:
void replaceTrack(ScriptExecutionContext&, RTCRtpSender&, RefPtr<MediaStreamTrack>&&, DOMPromiseDeferred<void>&&) final;
RTCRtpParameters getParameters() const final;
@@ -98,7 +102,7 @@
WeakPtr<LibWebRTCPeerConnectionBackend> m_peerConnectionBackend;
rtc::scoped_refptr<webrtc::RtpSenderInterface> m_rtcSender;
- Variant<std::nullptr_t, Ref<RealtimeOutgoingAudioSource>, Ref<RealtimeOutgoingVideoSource>> m_source;
+ Source m_source;
};
} // namespace WebCore
Added: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpTransceiverBackend.cpp (0 => 235809)
--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpTransceiverBackend.cpp (rev 0)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpTransceiverBackend.cpp 2018-09-07 21:57:47 UTC (rev 235809)
@@ -0,0 +1,78 @@
+/*
+ * Copyright (C) 2018 Apple Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ * 1. Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS'' AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON
+ * ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "config.h"
+#include "LibWebRTCRtpTransceiverBackend.h"
+
+#if ENABLE(WEB_RTC) && USE(LIBWEBRTC)
+
+#include "LibWebRTCRtpReceiverBackend.h"
+#include "LibWebRTCRtpSenderBackend.h"
+#include "LibWebRTCUtils.h"
+
+namespace WebCore {
+
+std::unique_ptr<LibWebRTCRtpReceiverBackend> LibWebRTCRtpTransceiverBackend::createReceiverBackend()
+{
+ return std::make_unique<LibWebRTCRtpReceiverBackend>(m_rtcTransceiver->receiver());
+}
+
+std::unique_ptr<LibWebRTCRtpSenderBackend> LibWebRTCRtpTransceiverBackend::createSenderBackend(LibWebRTCPeerConnectionBackend& backend, LibWebRTCRtpSenderBackend::Source&& source)
+{
+ return std::make_unique<LibWebRTCRtpSenderBackend>(backend, m_rtcTransceiver->sender(), WTFMove(source));
+}
+
+RTCRtpTransceiverDirection LibWebRTCRtpTransceiverBackend::direction() const
+{
+ return toRTCRtpTransceiverDirection(m_rtcTransceiver->direction());
+}
+
+std::optional<RTCRtpTransceiverDirection> LibWebRTCRtpTransceiverBackend::currentDirection() const
+{
+ auto value = m_rtcTransceiver->current_direction();
+ if (!value)
+ return std::nullopt;
+ return toRTCRtpTransceiverDirection(*value);
+}
+
+void LibWebRTCRtpTransceiverBackend::setDirection(RTCRtpTransceiverDirection direction)
+{
+ m_rtcTransceiver->SetDirection(fromRTCRtpTransceiverDirection(direction));
+}
+
+String LibWebRTCRtpTransceiverBackend::mid()
+{
+ if (auto mid = m_rtcTransceiver->mid())
+ return fromStdString(*mid);
+ return String { };
+}
+
+void LibWebRTCRtpTransceiverBackend::stop()
+{
+ m_rtcTransceiver->Stop();
+}
+
+} // namespace WebCore
+
+#endif // ENABLE(WEB_RTC) && USE(LIBWEBRTC)
Copied: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpTransceiverBackend.h (from rev 235808, trunk/Source/WebCore/Modules/mediastream/RTCRtpTransceiverBackend.h) (0 => 235809)
--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpTransceiverBackend.h (rev 0)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpTransceiverBackend.h 2018-09-07 21:57:47 UTC (rev 235809)
@@ -0,0 +1,69 @@
+/*
+ * Copyright (C) 2018 Apple Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ * 1. Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS'' AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON
+ * ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#pragma once
+
+#if ENABLE(WEB_RTC) && USE(LIBWEBRTC)
+
+#include "LibWebRTCMacros.h"
+#include "LibWebRTCRtpSenderBackend.h"
+#include "RTCRtpTransceiverBackend.h"
+
+#pragma GCC diagnostic push
+#pragma GCC diagnostic ignored "-Wunused-parameter"
+
+#include <webrtc/api/rtptransceiverinterface.h>
+#include <webrtc/rtc_base/scoped_ref_ptr.h>
+
+#pragma GCC diagnostic pop
+
+namespace WebCore {
+
+class LibWebRTCRtpReceiverBackend;
+
+class LibWebRTCRtpTransceiverBackend final : public RTCRtpTransceiverBackend {
+public:
+ explicit LibWebRTCRtpTransceiverBackend(rtc::scoped_refptr<webrtc::RtpTransceiverInterface>&& rtcTransceiver)
+ : m_rtcTransceiver(WTFMove(rtcTransceiver))
+ {
+ }
+
+ std::unique_ptr<LibWebRTCRtpReceiverBackend> createReceiverBackend();
+ std::unique_ptr<LibWebRTCRtpSenderBackend> createSenderBackend(LibWebRTCPeerConnectionBackend&, LibWebRTCRtpSenderBackend::Source&&);
+
+ webrtc::RtpTransceiverInterface* rtcTransceiver() { return m_rtcTransceiver.get(); }
+
+private:
+ RTCRtpTransceiverDirection direction() const final;
+ std::optional<RTCRtpTransceiverDirection> currentDirection() const final;
+ void setDirection(RTCRtpTransceiverDirection) final;
+ String mid() final;
+ void stop() final;
+
+ rtc::scoped_refptr<webrtc::RtpTransceiverInterface> m_rtcTransceiver;
+};
+
+} // namespace WebCore
+
+#endif // ENABLE(WEB_RTC) && USE(LIBWEBRTC)
Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCUtils.cpp (235808 => 235809)
--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCUtils.cpp 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCUtils.cpp 2018-09-07 21:57:47 UTC (rev 235809)
@@ -28,10 +28,18 @@
#if USE(LIBWEBRTC)
#include "LibWebRTCMacros.h"
+#include "RTCPeerConnection.h"
#include "RTCRtpParameters.h"
-#include <webrtc/api/rtpparameters.h>
#include <wtf/text/WTFString.h>
+#pragma GCC diagnostic push
+#pragma GCC diagnostic ignored "-Wunused-parameter"
+
+#include <webrtc/api/rtpparameters.h>
+#include <webrtc/api/rtptransceiverinterface.h>
+
+#pragma GCC diagnostic pop
+
namespace WebCore {
static inline RTCRtpParameters::EncodingParameters toRTCEncodingParameters(const webrtc::RtpEncodingParameters& rtcParameters)
@@ -177,6 +185,41 @@
return rtcParameters;
}
+RTCRtpTransceiverDirection toRTCRtpTransceiverDirection(webrtc::RtpTransceiverDirection rtcDirection)
+{
+ switch (rtcDirection) {
+ case webrtc::RtpTransceiverDirection::kSendRecv:
+ return RTCRtpTransceiverDirection::Sendrecv;
+ case webrtc::RtpTransceiverDirection::kSendOnly:
+ return RTCRtpTransceiverDirection::Sendonly;
+ case webrtc::RtpTransceiverDirection::kRecvOnly:
+ return RTCRtpTransceiverDirection::Recvonly;
+ case webrtc::RtpTransceiverDirection::kInactive:
+ return RTCRtpTransceiverDirection::Inactive;
+ };
+}
+
+webrtc::RtpTransceiverDirection fromRTCRtpTransceiverDirection(RTCRtpTransceiverDirection direction)
+{
+ switch (direction) {
+ case RTCRtpTransceiverDirection::Sendrecv:
+ return webrtc::RtpTransceiverDirection::kSendRecv;
+ case RTCRtpTransceiverDirection::Sendonly:
+ return webrtc::RtpTransceiverDirection::kSendOnly;
+ case RTCRtpTransceiverDirection::Recvonly:
+ return webrtc::RtpTransceiverDirection::kRecvOnly;
+ case RTCRtpTransceiverDirection::Inactive:
+ return webrtc::RtpTransceiverDirection::kInactive;
+ };
+}
+
+webrtc::RtpTransceiverInit fromRtpTransceiverInit(const RTCRtpTransceiverInit& init)
+{
+ webrtc::RtpTransceiverInit rtcInit;
+ rtcInit.direction = fromRTCRtpTransceiverDirection(init.direction);
+ return rtcInit;
+}
+
}; // namespace WebCore
#endif // USE(LIBWEBRTC)
Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCUtils.h (235808 => 235809)
--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCUtils.h 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCUtils.h 2018-09-07 21:57:47 UTC (rev 235809)
@@ -30,15 +30,25 @@
namespace webrtc {
struct RtpParameters;
+struct RtpTransceiverInit;
+
+enum class RtpTransceiverDirection;
}
namespace WebCore {
struct RTCRtpParameters;
+struct RTCRtpTransceiverInit;
+enum class RTCRtpTransceiverDirection;
+
RTCRtpParameters toRTCRtpParameters(const webrtc::RtpParameters&);
webrtc::RtpParameters fromRTCRtpParameters(const RTCRtpParameters&);
+RTCRtpTransceiverDirection toRTCRtpTransceiverDirection(webrtc::RtpTransceiverDirection);
+webrtc::RtpTransceiverDirection fromRTCRtpTransceiverDirection(RTCRtpTransceiverDirection);
+webrtc::RtpTransceiverInit fromRtpTransceiverInit(const RTCRtpTransceiverInit&);
+
inline String fromStdString(const std::string& value)
{
return String::fromUTF8(value.data(), value.length());
Modified: trunk/Source/WebCore/WebCore.xcodeproj/project.pbxproj (235808 => 235809)
--- trunk/Source/WebCore/WebCore.xcodeproj/project.pbxproj 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/Source/WebCore/WebCore.xcodeproj/project.pbxproj 2018-09-07 21:57:47 UTC (rev 235809)
@@ -1103,6 +1103,7 @@
4186BD3E213EE3400001826F /* LibWebRTCUtils.cpp in Sources */ = {isa = PBXBuildFile; fileRef = 41D1A049213EDDFD0063FB6B /* LibWebRTCUtils.cpp */; };
4186BD3F213EE3430001826F /* LibWebRTCRtpSenderBackend.cpp in Sources */ = {isa = PBXBuildFile; fileRef = 41D1A04B213EDDFE0063FB6B /* LibWebRTCRtpSenderBackend.cpp */; };
4186BD40213EE3450001826F /* LibWebRTCRtpReceiverBackend.cpp in Sources */ = {isa = PBXBuildFile; fileRef = 41D1A04A213EDDFE0063FB6B /* LibWebRTCRtpReceiverBackend.cpp */; };
+ 4186BD4E2140B9E80001826F /* LibWebRTCRtpTransceiverBackend.cpp in Sources */ = {isa = PBXBuildFile; fileRef = 4186BD4D2140B9E80001826F /* LibWebRTCRtpTransceiverBackend.cpp */; };
41885B9311B6FDA6003383BB /* FormSubmission.h in Headers */ = {isa = PBXBuildFile; fileRef = 41885B9111B6FDA6003383BB /* FormSubmission.h */; settings = {ATTRIBUTES = (Private, ); }; };
418A06D0133C04D500CD379C /* EventDispatcher.h in Headers */ = {isa = PBXBuildFile; fileRef = 418A06CE133C04D500CD379C /* EventDispatcher.h */; };
418F88050FF957AF0080F045 /* JSAbstractWorker.h in Headers */ = {isa = PBXBuildFile; fileRef = 418F88030FF957AE0080F045 /* JSAbstractWorker.h */; };
@@ -7288,6 +7289,8 @@
4186BD3B213EDE380001826F /* LibWebRTCRtpReceiverBackend.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; name = LibWebRTCRtpReceiverBackend.h; path = libwebrtc/LibWebRTCRtpReceiverBackend.h; sourceTree = "<group>"; };
4186BD3D213EDE390001826F /* LibWebRTCRtpSenderBackend.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; name = LibWebRTCRtpSenderBackend.h; path = libwebrtc/LibWebRTCRtpSenderBackend.h; sourceTree = "<group>"; };
4186BD46214072B60001826F /* RTCRtpTransceiverBackend.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = RTCRtpTransceiverBackend.h; sourceTree = "<group>"; };
+ 4186BD4B2140A8050001826F /* LibWebRTCRtpTransceiverBackend.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = LibWebRTCRtpTransceiverBackend.h; path = libwebrtc/LibWebRTCRtpTransceiverBackend.h; sourceTree = "<group>"; };
+ 4186BD4D2140B9E80001826F /* LibWebRTCRtpTransceiverBackend.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = LibWebRTCRtpTransceiverBackend.cpp; path = libwebrtc/LibWebRTCRtpTransceiverBackend.cpp; sourceTree = "<group>"; };
41885B9111B6FDA6003383BB /* FormSubmission.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = FormSubmission.h; sourceTree = "<group>"; };
41885B9211B6FDA6003383BB /* FormSubmission.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = FormSubmission.cpp; sourceTree = "<group>"; };
418A06CE133C04D500CD379C /* EventDispatcher.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = EventDispatcher.h; sourceTree = "<group>"; };
@@ -16894,6 +16897,8 @@
4186BD3B213EDE380001826F /* LibWebRTCRtpReceiverBackend.h */,
41D1A04B213EDDFE0063FB6B /* LibWebRTCRtpSenderBackend.cpp */,
4186BD3D213EDE390001826F /* LibWebRTCRtpSenderBackend.h */,
+ 4186BD4D2140B9E80001826F /* LibWebRTCRtpTransceiverBackend.cpp */,
+ 4186BD4B2140A8050001826F /* LibWebRTCRtpTransceiverBackend.h */,
41D28D0B2139E01D00F4206F /* LibWebRTCStatsCollector.cpp */,
41D28D0C2139E01E00F4206F /* LibWebRTCStatsCollector.h */,
41D1A049213EDDFD0063FB6B /* LibWebRTCUtils.cpp */,
@@ -28781,7 +28786,6 @@
D6489D26166FFCF1007C031B /* JSHTMLTemplateElement.h in Headers */,
A80E7E9D0A1A83E3007FB8C5 /* JSHTMLTextAreaElement.h in Headers */,
83E359A21BB1031D002CEB98 /* JSHTMLTimeElement.h in Headers */,
- E42050172141901B0066EF3B /* ProcessWarming.h in Headers */,
A80E7B0C0A19D606007FB8C5 /* JSHTMLTitleElement.h in Headers */,
070756D414239A4F00414161 /* JSHTMLTrackElement.h in Headers */,
1A85B2110A1B258700D8C87C /* JSHTMLUListElement.h in Headers */,
@@ -29789,6 +29793,7 @@
B71FE6DF11091CB300DAEF77 /* PrintContext.h in Headers */,
51F645D51FECDBCE00B54DED /* Process.h in Headers */,
A8EA7EBC0A1945D000A8EF5F /* ProcessingInstruction.h in Headers */,
+ E42050172141901B0066EF3B /* ProcessWarming.h in Headers */,
E44613EC0CD681B500FADA75 /* ProgressEvent.h in Headers */,
A715E653134BBBEC00D8E713 /* ProgressShadowElement.h in Headers */,
1A2A68240B5BEDE70002A480 /* ProgressTracker.h in Headers */,
@@ -31525,6 +31530,7 @@
417612B11E3A994000C3D81D /* LibWebRTCPeerConnectionBackend.cpp in Sources */,
4186BD40213EE3450001826F /* LibWebRTCRtpReceiverBackend.cpp in Sources */,
4186BD3F213EE3430001826F /* LibWebRTCRtpSenderBackend.cpp in Sources */,
+ 4186BD4E2140B9E80001826F /* LibWebRTCRtpTransceiverBackend.cpp in Sources */,
41D28D0D2139E05800F4206F /* LibWebRTCStatsCollector.cpp in Sources */,
4186BD3E213EE3400001826F /* LibWebRTCUtils.cpp in Sources */,
9759E93E14EF1CF80026A2DD /* LoadableTextTrack.cpp in Sources */,
Modified: trunk/Source/WebCore/platform/mediastream/RealtimeIncomingAudioSource.cpp (235808 => 235809)
--- trunk/Source/WebCore/platform/mediastream/RealtimeIncomingAudioSource.cpp 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/Source/WebCore/platform/mediastream/RealtimeIncomingAudioSource.cpp 2018-09-07 21:57:47 UTC (rev 235809)
@@ -41,6 +41,7 @@
: RealtimeMediaSource(WTFMove(audioTrackId), RealtimeMediaSource::Type::Audio, String())
, m_audioTrack(WTFMove(audioTrack))
{
+ setName("remote audio");
notifyMutedChange(!m_audioTrack);
}
Modified: trunk/Source/WebCore/platform/mediastream/RealtimeIncomingVideoSource.cpp (235808 => 235809)
--- trunk/Source/WebCore/platform/mediastream/RealtimeIncomingVideoSource.cpp 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/Source/WebCore/platform/mediastream/RealtimeIncomingVideoSource.cpp 2018-09-07 21:57:47 UTC (rev 235809)
@@ -41,6 +41,7 @@
: RealtimeMediaSource(WTFMove(videoTrackId), RealtimeMediaSource::Type::Video, String())
, m_videoTrack(WTFMove(videoTrack))
{
+ setName("remote video");
m_currentSettings.setWidth(640);
m_currentSettings.setHeight(480);
notifyMutedChange(!m_videoTrack);
Modified: trunk/Source/WebCore/platform/mediastream/RealtimeMediaSource.h (235808 => 235809)
--- trunk/Source/WebCore/platform/mediastream/RealtimeMediaSource.h 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/Source/WebCore/platform/mediastream/RealtimeMediaSource.h 2018-09-07 21:57:47 UTC (rev 235809)
@@ -158,7 +158,7 @@
virtual void setInterrupted(bool, bool);
const String& name() const { return m_name; }
- void setName(const String& name) { m_name = name; }
+ void setName(String&& name) { m_name = WTFMove(name); }
unsigned fitnessScore() const { return m_fitnessScore; }
Modified: trunk/Source/WebCore/testing/Internals.cpp (235808 => 235809)
--- trunk/Source/WebCore/testing/Internals.cpp 2018-09-07 21:43:35 UTC (rev 235808)
+++ trunk/Source/WebCore/testing/Internals.cpp 2018-09-07 21:57:47 UTC (rev 235809)
@@ -1386,6 +1386,9 @@
void Internals::useMockRTCPeerConnectionFactory(const String& testCase)
{
+ // FIXME: We should upgrade mocks to support unified plan APIs, until then use plan B in tests using mock.
+
+ ASSERT(!RuntimeEnabledFeatures::sharedFeatures().webRTCUnifiedPlanEnabled());
if (!LibWebRTCProvider::webRTCAvailable())
return;
