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[WebRTC] Add support for legacy Offer options
https://bugs.webkit.org/show_bug.cgi?id=256707 <rdar://problem/109569164> Reviewed by Youenn Fablet. Some websites still rely on the legacy offer options, despite the spec recommendation of using transceivers, so bring back those options, behind a runtime WebPreference, disabled by default. Covered by new tests backported from WPT. Upstream commit: web-platform-tests/wpt@9c29d4e * LayoutTests/TestExpectations: * LayoutTests/platform/glib/TestExpectations: * Source/WTF/Scripts/Preferences/UnifiedWebPreferences.yaml: * Source/WebCore/Modules/mediastream/PeerConnectionBackend.cpp: (WebCore::PeerConnectionBackend::addTransceiver): * Source/WebCore/Modules/mediastream/PeerConnectionBackend.h: * Source/WebCore/Modules/mediastream/RTCOfferOptions.h: * Source/WebCore/Modules/mediastream/RTCOfferOptions.idl: * Source/WebCore/Modules/mediastream/RTCPeerConnection.cpp: (WebCore::RTCPeerConnection::addReceiveOnlyTransceiver): (WebCore::RTCPeerConnection::createOffer): * Source/WebCore/Modules/mediastream/RTCPeerConnection.h: * Source/WebCore/Modules/mediastream/gstreamer/GStreamerMediaEndpoint.cpp: (WebCore::GStreamerMediaEndpoint::createTransceiverBackends): (WebCore::GStreamerMediaEndpoint::addTransceiver): (WebCore::GStreamerMediaEndpoint::onNegotiationNeeded): * Source/WebCore/Modules/mediastream/gstreamer/GStreamerMediaEndpoint.h: * Source/WebCore/Modules/mediastream/gstreamer/GStreamerPeerConnectionBackend.cpp: (WebCore::GStreamerPeerConnectionBackend::addTransceiverFromTrackOrKind): (WebCore::GStreamerPeerConnectionBackend::addTransceiver): * Source/WebCore/Modules/mediastream/gstreamer/GStreamerPeerConnectionBackend.h: * Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp: (WebCore::LibWebRTCMediaEndpoint::createTransceiverBackends): (WebCore::LibWebRTCMediaEndpoint::addTransceiver): (WebCore::LibWebRTCMediaEndpoint::OnNegotiationNeededEvent): * Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h: * Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp: (WebCore::LibWebRTCPeerConnectionBackend::addTransceiverFromTrackOrKind): (WebCore::LibWebRTCPeerConnectionBackend::addTransceiver): * Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h: Canonical link: https://commits.webkit.org/283089@main
1 parent b7683c4 commit e628ffd

27 files changed

Lines changed: 358 additions & 66 deletions

‎LayoutTests/TestExpectations‎

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@@ -245,6 +245,16 @@ imported/w3c/web-platform-tests/video-rvfc [ Skip ]
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fast/mediastream/getUserMedia-rvfc.html [ Skip ]
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webrtc/peerConnection-rvfc.html [ Skip ]
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248+
# Our implementation of pc.setLocalDescription() is not fully spec compliant, steps 4.2 and 4.3 are
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# not implemented.
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imported/w3c/web-platform-tests/webrtc/legacy/munge-dont.html [ Failure ]
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# The addstream event is not supported anymore.
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imported/w3c/web-platform-tests/webrtc/legacy/onaddstream.https.html [ Skip ]
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# The non-promise based createOffer/createAnswer functions are not supported anymore.
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imported/w3c/web-platform-tests/webrtc/legacy/simplecall_callbacks.https.html [ Skip ]
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# This test only makes sense on Mac
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fast/attachment/attachment-subtitle-resize.html [ Skip ]
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‎LayoutTests/imported/w3c/resources/import-expectations.json‎

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"web-platform-tests/webrtc": "import",
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"web-platform-tests/webrtc-extensions": "import",
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"web-platform-tests/webrtc-priority": "import",
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"web-platform-tests/webrtc/legacy": "skip",
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"web-platform-tests/websockets": "import",
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"web-platform-tests/webstorage": "import",
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"web-platform-tests/webstorage/": "import",

‎LayoutTests/imported/w3c/web-platform-tests/webrtc/legacy/RTCPeerConnection-createOffer-offerToReceive-expected.txt‎

Lines changed: 14 additions & 14 deletions
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@@ -1,20 +1,20 @@
11

2-
FAIL createOffer() with offerToReceiveAudio should add audio line to all subsequent created offers assert_equals: Expect created offer to have audio line expected 1 but got 0
3-
FAIL createOffer() with offerToReceiveVideo should add video line to all subsequent created offers assert_equals: Expect created offer to have video line expected 1 but got 0
4-
FAIL createOffer() with offerToReceiveAudio:true, then with offerToReceiveVideo:true, should have result offer with both audio and video line assert_equals: Expect audio line to be found in created offer expected 1 but got 0
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PASS createOffer() with offerToReceiveAudio should add audio line to all subsequent created offers
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PASS createOffer() with offerToReceiveVideo should add video line to all subsequent created offers
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PASS createOffer() with offerToReceiveAudio:true, then with offerToReceiveVideo:true, should have result offer with both audio and video line
55
PASS createOffer() with offerToReceiveAudio set to false should not create a transceiver
6-
FAIL createOffer() with offerToReceiveAudio should create a "recvonly" transceiver assert_equals: Expect pc to have one transceiver expected 1 but got 0
7-
FAIL offerToReceiveAudio option should be ignored if a non-stopped "recvonly" transceiver exists assert_equals: Expect pc to have one transceiver expected 1 but got 0
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PASS createOffer() with offerToReceiveAudio should create a "recvonly" transceiver
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PASS offerToReceiveAudio option should be ignored if a non-stopped "recvonly" transceiver exists
88
PASS offerToReceiveAudio option should be ignored if a non-stopped "sendrecv" transceiver exists
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FAIL offerToReceiveAudio set to false with a track should create a "sendonly" transceiver assert_equals: Expect transceiver to have "sendonly" direction expected "sendonly" but got "sendrecv"
10-
FAIL offerToReceiveAudio set to false with a "recvonly" transceiver should change the direction to "inactive" assert_equals: Expect transceiver to have "inactive" direction expected "inactive" but got "recvonly"
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FAIL subsequent offerToReceiveAudio set to false with a track should change the direction to "sendonly" assert_equals: Expect transceiver to have "sendonly" direction expected "sendonly" but got "sendrecv"
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PASS offerToReceiveAudio set to false with a track should create a "sendonly" transceiver
10+
PASS offerToReceiveAudio set to false with a "recvonly" transceiver should change the direction to "inactive"
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PASS subsequent offerToReceiveAudio set to false with a track should change the direction to "sendonly"
1212
PASS createOffer() with offerToReceiveVideo set to false should not create a transceiver
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FAIL createOffer() with offerToReceiveVideo should create a "recvonly" transceiver assert_equals: Expect pc to have one transceiver expected 1 but got 0
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FAIL offerToReceiveVideo option should be ignored if a non-stopped "recvonly" transceiver exists assert_equals: Expect pc to have one transceiver expected 1 but got 0
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PASS createOffer() with offerToReceiveVideo should create a "recvonly" transceiver
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PASS offerToReceiveVideo option should be ignored if a non-stopped "recvonly" transceiver exists
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PASS offerToReceiveVideo option should be ignored if a non-stopped "sendrecv" transceiver exists
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FAIL offerToReceiveVideo set to false with a track should create a "sendonly" transceiver assert_equals: Expect transceiver to have "sendonly" direction expected "sendonly" but got "sendrecv"
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FAIL offerToReceiveVideo set to false with a "recvonly" transceiver should change the direction to "inactive" assert_equals: Expect transceiver to have "inactive" direction expected "inactive" but got "recvonly"
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FAIL subsequent offerToReceiveVideo set to false with a track should change the direction to "sendonly" assert_equals: Expect transceiver to have "sendonly" direction expected "sendonly" but got "sendrecv"
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FAIL offerToReceiveAudio and Video should create two "recvonly" transceivers assert_equals: Expect pc to have two transceivers expected 2 but got 0
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PASS offerToReceiveVideo set to false with a track should create a "sendonly" transceiver
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PASS offerToReceiveVideo set to false with a "recvonly" transceiver should change the direction to "inactive"
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PASS subsequent offerToReceiveVideo set to false with a track should change the direction to "sendonly"
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PASS offerToReceiveAudio and Video should create two "recvonly" transceivers
2020

‎LayoutTests/imported/w3c/web-platform-tests/webrtc/legacy/RTCPeerConnection-createOffer-offerToReceive.html‎

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<!doctype html>
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<!doctype html><!-- webkit-test-runner [ LegacyWebRTCOfferOptionsEnabled=true ] -->
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<meta charset=utf-8>
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<title>Test legacy offerToReceiveAudio/Video options</title>
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<link rel="help" href="https://w3c.github.io/webrtc-pc/#legacy-configuration-extensions">

‎LayoutTests/imported/w3c/web-platform-tests/webrtc/legacy/RTCRtpTransceiver-with-OfferToReceive-options.https-expected.txt‎

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11

22
PASS checkAddTransceiverWithStream
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FAIL checkAddTransceiverWithOfferToReceiveAudio assert_equals: expected "[{currentDirection:null,direction:\"recvonly\",mid:null,receiver:{track:{kind:\"audio\"}},sender:{track:null},stopped:false}]" but got "[]"
4-
FAIL checkAddTransceiverWithOfferToReceiveVideo assert_equals: expected "[{currentDirection:null,direction:\"recvonly\",mid:null,receiver:{track:{kind:\"video\"}},sender:{track:null},stopped:false}]" but got "[]"
5-
FAIL checkAddTransceiverWithOfferToReceiveBoth assert_equals: expected "[{currentDirection:null,direction:\"recvonly\",mid:null,receiver:{track:{kind:\"audio\"}},sender:{track:null},stopped:false},{currentDirection:null,direction:\"recvonly\",mid:null,receiver:{track:{kind:\"video\"}},sender:{track:null},stopped:false}]" but got "[]"
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PASS checkAddTransceiverWithOfferToReceiveAudio
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PASS checkAddTransceiverWithOfferToReceiveVideo
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PASS checkAddTransceiverWithOfferToReceiveBoth
66

‎LayoutTests/imported/w3c/web-platform-tests/webrtc/legacy/RTCRtpTransceiver-with-OfferToReceive-options.https.html‎

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<!doctype html>
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<!doctype html><!-- webkit-test-runner [ LegacyWebRTCOfferOptionsEnabled=true ] -->
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<meta charset=utf-8>
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<title>RTCRtpTransceiver with OfferToReceive legacy options</title>
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<script src="/resources/testharness.js"></script>
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<script src="/resources/testharnessreport.js"></script>
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<script src=/resources/testdriver.js></script>
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<script src=/resources/testdriver-vendor.js></script>
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<script src='/mediacapture-streams/permission-helper.js'></script>
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<script src="../RTCPeerConnection-helper.js"></script>
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<script>
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'use strict';
@@ -37,7 +40,7 @@
3740
const checkAddTransceiverWithStream = async t => {
3841
const pc = new RTCPeerConnection();
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t.add_cleanup(() => pc.close());
40-
43+
await setMediaPermission();
4144
const audioStream = await navigator.mediaDevices.getUserMedia({audio: true});
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const videoStream = await navigator.mediaDevices.getUserMedia({video: true});
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t.add_cleanup(() => stopTracks(audioStream, videoStream));

‎LayoutTests/imported/w3c/web-platform-tests/webrtc/legacy/munge-dont-expected.txt‎

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PASS RTCSessionDescription.type is read-only
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PASS RTCSessionDescription.sdp is read-only
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PASS RTCIceCandidate.candidate is read-only
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PASS Rejects SDP munging between createOffer and setLocalDescription
6+
PASS Rejects SDP munging between createAnswer and setLocalDescription
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‎LayoutTests/imported/w3c/web-platform-tests/webrtc/legacy/munge-dont.html‎

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<!doctype html>
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<meta charset=utf-8>
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<meta name="timeout" content="long">
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<title>SDP munging is a bad idea</title>
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<script src="/resources/testharness.js"></script>
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<script src="/resources/testharnessreport.js"></script>
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<script>
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'use strict';
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const sdp = `v=0
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o=- 0 3 IN IP4 127.0.0.1
12+
s=-
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t=0 0
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m=video 9 UDP/TLS/RTP/SAVPF 100
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c=IN IP4 0.0.0.0
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a=rtcp-mux
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a=sendonly
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a=mid:video
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a=rtpmap:100 VP8/90000
20+
a=fmtp:100 max-fr=30;max-fs=3600
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a=fingerprint:sha-256 A7:24:72:CA:6E:02:55:39:BA:66:DF:6E:CC:4C:D8:B0:1A:BF:1A:56:65:7D:F4:03:AD:7E:77:43:2A:29:EC:93
22+
a=ice-ufrag:ETEn
23+
a=ice-pwd:OtSK0WpNtpUjkY4+86js7Z/l
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`;
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const candidateString = 'candidate:1905690388 1 udp 2113937151 192.168.0.1 58041 typ host generation 0 ufrag thC8';
26+
27+
// See https://bugs.chromium.org/p/chromium/issues/detail?id=662898
28+
// and https://bugs.chromium.org/p/chromium/issues/detail?id=823036
29+
// for why neither of these is feasible to enforce.
30+
31+
// Note that this does not restrict creating a
32+
// new RTCSessionDescription with a modified copy.
33+
test(() => {
34+
const desc = new RTCSessionDescription({
35+
type: 'offer',
36+
sdp,
37+
});
38+
assert_throws_js(TypeError, () => {
39+
desc.type = 'answer';
40+
});
41+
}, 'RTCSessionDescription.type is read-only');
42+
43+
test(() => {
44+
const desc = new RTCSessionDescription({
45+
type: 'offer',
46+
sdp,
47+
});
48+
assert_throws_js(TypeError, () => {
49+
desc.sdp += 'a=dont-modify-me\r\n';
50+
});
51+
}, 'RTCSessionDescription.sdp is read-only');
52+
53+
test(() => {
54+
const candidate = new RTCIceCandidate({
55+
sdpMid: '0',
56+
candidate: candidateString,
57+
});
58+
assert_throws_js(TypeError, () => {
59+
candidate.candidate += ' myattribute value';
60+
});
61+
}, 'RTCIceCandidate.candidate is read-only');
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// https://w3c.github.io/webrtc-pc/#dom-peerconnection-setlocaldescription
64+
// If type is "offer", and sdp is not the empty string and not equal to
65+
// connection.[[LastCreatedOffer]], then return a promise rejected with a
66+
// newly created InvalidModificationError and abort these steps.
67+
promise_test(async t => {
68+
const pc = new RTCPeerConnection();
69+
t.add_cleanup(() => pc.close());
70+
pc.addTransceiver('audio');
71+
const offer = await pc.createOffer();
72+
return promise_rejects_dom(t, 'InvalidModificationError',
73+
pc.setLocalDescription({type: 'offer', sdp: offer.sdp + 'a=munging-is-not-good\r\n'}));
74+
}, 'Rejects SDP munging between createOffer and setLocalDescription');
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76+
// If type is "answer" or "pranswer", and sdp is not the empty string and not equal to
77+
// connection.[[LastCreatedAnswer]], then return a promise rejected with a
78+
// newly created InvalidModificationError and abort these steps.
79+
promise_test(async t => {
80+
const pc = new RTCPeerConnection();
81+
t.add_cleanup(() => pc.close());
82+
await pc.setRemoteDescription({type: 'offer', sdp});
83+
84+
const answer = await pc.createAnswer();
85+
return promise_rejects_dom(t, 'InvalidModificationError',
86+
pc.setLocalDescription({type: 'answer', sdp: answer.sdp + 'a=munging-is-not-good\r\n'}));
87+
}, 'Rejects SDP munging between createAnswer and setLocalDescription');
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</script>

‎LayoutTests/imported/w3c/web-platform-tests/webrtc/legacy/onaddstream.https.html‎

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<title>onaddstream tests</title>
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<script src="/resources/testharness.js"></script>
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<script src="/resources/testharnessreport.js"></script>
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<script src=/resources/testdriver.js></script>
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<script src=/resources/testdriver-vendor.js></script>
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<script src='/mediacapture-streams/permission-helper.js'></script>
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<script>
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'use strict';
811

@@ -90,6 +93,7 @@
9093
promise_test(async t => {
9194
const pc1 = new RTCPeerConnection();
9295
t.add_cleanup(() => pc1.close());
96+
await setMediaPermission();
9397
const stream1 = await navigator.mediaDevices.getUserMedia({audio: true, video: true});
9498
t.add_cleanup(() => stopTracks(stream1));
9599
const audio1 = stream1.getAudioTracks()[0];

‎LayoutTests/imported/w3c/web-platform-tests/webrtc/legacy/simplecall_callbacks.https.html‎

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<!doctype html>
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<html>
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<head>
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<meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
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<title>RTCPeerConnection Connection Test</title>
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<script src="../RTCPeerConnection-helper.js"></script>
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</head>
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<body>
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<div id="log"></div>
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<div>
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<video id="local-view" muted autoplay="autoplay"></video>
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<video id="remote-view" muted autoplay="autoplay"></video>
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</div>
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<!-- These files are in place when executing on W3C. -->
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<script src="/resources/testharness.js"></script>
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<script src="/resources/testharnessreport.js"></script>
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<script type="text/javascript">
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var test = async_test('Can set up a basic WebRTC call.');
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var gFirstConnection = null;
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var gSecondConnection = null;
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// if the remote video gets video data that implies the negotiation
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// as well as the ICE and DTLS connection are up.
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document.getElementById('remote-view')
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.addEventListener('loadedmetadata', function() {
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// Call negotiated: done.
29+
test.done();
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});
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function getNoiseStreamOkCallback(localStream) {
33+
gFirstConnection = new RTCPeerConnection(null);
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test.add_cleanup(() => gFirstConnection.close());
35+
gFirstConnection.onicecandidate = onIceCandidateToFirst;
36+
localStream.getTracks().forEach(function(track) {
37+
gFirstConnection.addTrack(track, localStream);
38+
});
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gFirstConnection.createOffer(onOfferCreated, failed('createOffer'));
40+
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var videoTag = document.getElementById('local-view');
42+
videoTag.srcObject = localStream;
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};
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45+
var onOfferCreated = test.step_func(function(offer) {
46+
gFirstConnection.setLocalDescription(offer);
47+
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// This would normally go across the application's signaling solution.
49+
// In our case, the "signaling" is to call this function.
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receiveCall(offer.sdp);
51+
});
52+
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function receiveCall(offerSdp) {
54+
gSecondConnection = new RTCPeerConnection(null);
55+
test.add_cleanup(() => gSecondConnection.close());
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gSecondConnection.onicecandidate = onIceCandidateToSecond;
57+
gSecondConnection.ontrack = onRemoteTrack;
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var parsedOffer = new RTCSessionDescription({ type: 'offer',
60+
sdp: offerSdp });
61+
gSecondConnection.setRemoteDescription(parsedOffer);
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gSecondConnection.createAnswer(onAnswerCreated,
64+
failed('createAnswer'));
65+
};
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var onAnswerCreated = test.step_func(function(answer) {
68+
gSecondConnection.setLocalDescription(answer);
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// Similarly, this would go over the application's signaling solution.
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handleAnswer(answer.sdp);
72+
});
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function handleAnswer(answerSdp) {
75+
var parsedAnswer = new RTCSessionDescription({ type: 'answer',
76+
sdp: answerSdp });
77+
gFirstConnection.setRemoteDescription(parsedAnswer);
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};
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var onIceCandidateToFirst = test.step_func(function(event) {
81+
gSecondConnection.addIceCandidate(event.candidate);
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});
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var onIceCandidateToSecond = test.step_func(function(event) {
85+
gFirstConnection.addIceCandidate(event.candidate);
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});
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var onRemoteTrack = test.step_func(function(event) {
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var videoTag = document.getElementById('remote-view');
90+
if (!videoTag.srcObject) {
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videoTag.srcObject = event.streams[0];
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}
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});
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// Returns a suitable error callback.
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function failed(function_name) {
97+
return test.unreached_func('WebRTC called error callback for ' + function_name);
98+
}
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// This function starts the test.
101+
test.step(function() {
102+
getNoiseStream({ video: true, audio: true })
103+
.then(test.step_func(getNoiseStreamOkCallback), failed('getNoiseStream'));
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});
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</script>
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</body>
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</html>

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