Diff
Modified: trunk/LayoutTests/ChangeLog (239251 => 239252)
--- trunk/LayoutTests/ChangeLog 2018-12-15 10:03:19 UTC (rev 239251)
+++ trunk/LayoutTests/ChangeLog 2018-12-15 17:40:26 UTC (rev 239252)
@@ -1,3 +1,13 @@
+2018-12-15 Youenn Fablet <[email protected]>
+
+ Make RTCRtpSender.setParameters to activate specific encodings
+ https://bugs.webkit.org/show_bug.cgi?id=192732
+
+ Reviewed by Eric Carlson.
+
+ * webrtc/video-expected.txt:
+ * webrtc/video.html:
+
2018-12-15 Nikita Vasilyev <[email protected]>
Web Inspector: Styles: toggling selected properties may cause data corruption
Modified: trunk/LayoutTests/webrtc/video-expected.txt (239251 => 239252)
--- trunk/LayoutTests/webrtc/video-expected.txt 2018-12-15 10:03:19 UTC (rev 239251)
+++ trunk/LayoutTests/webrtc/video-expected.txt 2018-12-15 17:40:26 UTC (rev 239252)
@@ -1,4 +1,6 @@
PASS Basic video exchange
+PASS Call setParameters to disable sending a given encoding
+PASS Call setParameters to reenable sending a given encoding
Modified: trunk/LayoutTests/webrtc/video.html (239251 => 239252)
--- trunk/LayoutTests/webrtc/video.html 2018-12-15 10:03:19 UTC (rev 239251)
+++ trunk/LayoutTests/webrtc/video.html 2018-12-15 17:40:26 UTC (rev 239252)
@@ -39,37 +39,103 @@
assert_true(data[index + 2] < 100);
}
-promise_test((test) => {
+var pc1, pc2;
+promise_test(async (test) => {
if (window.testRunner)
testRunner.setUserMediaPermission(true);
- return navigator.mediaDevices.getUserMedia({video: {advanced: [{width:{min:640}}, {height:{min:480} } ]}}).then((stream) => {
- if (window.internals)
- assert_true(internals.pageMediaState().includes('HasActiveVideoCaptureDevice'), "Unexpected HasActiveVideoCaptureDevice");
- return new Promise((resolve, reject) => {
- createConnections((firstConnection) => {
- var track = stream.getVideoTracks()[0];
- firstConnection.addTrack(stream.getVideoTracks()[0], stream);
- }, (secondConnection) => {
- secondConnection._ontrack_ = (trackEvent) => {
- assert_true(trackEvent.track instanceof MediaStreamTrack);
- assert_true(trackEvent.receiver instanceof RTCRtpReceiver);
- assert_true(Array.isArray(trackEvent.streams), "Array.isArray() should return true");
- assert_true(Object.isFrozen(trackEvent.streams), "Object.isFrozen() should return true");
- assert_equals(trackEvent.track.id, stream.getVideoTracks()[0].id);
- assert_equals(trackEvent.track, trackEvent.streams[0].getVideoTracks()[0]);
- resolve(trackEvent.streams[0]);
- };
- });
- setTimeout(() => reject("Test timed out"), 5000);
+ const localStream = await navigator.mediaDevices.getUserMedia({video: {advanced: [{width:{min:640}}, {height:{min:480} } ]}});
+ if (window.internals)
+ assert_true(internals.pageMediaState().includes('HasActiveVideoCaptureDevice'), "Unexpected HasActiveVideoCaptureDevice");
+ const stream = await new Promise((resolve, reject) => {
+ createConnections((firstConnection) => {
+ pc1 = firstConnection;
+ firstConnection.addTrack(localStream.getVideoTracks()[0], localStream);
+ }, (secondConnection) => {
+ pc2 = secondConnection;
+ secondConnection._ontrack_ = (trackEvent) => {
+ assert_true(trackEvent.track instanceof MediaStreamTrack);
+ assert_true(trackEvent.receiver instanceof RTCRtpReceiver);
+ assert_true(Array.isArray(trackEvent.streams), "Array.isArray() should return true");
+ assert_true(Object.isFrozen(trackEvent.streams), "Object.isFrozen() should return true");
+ assert_equals(trackEvent.track.id, localStream.getVideoTracks()[0].id);
+ assert_equals(trackEvent.track, trackEvent.streams[0].getVideoTracks()[0]);
+ resolve(trackEvent.streams[0]);
+ };
});
- }).then((stream) => {
- video.srcObject = stream;
- return video.play();
- }).then(test.step_func(() => {
- testImage();
- }));
+ setTimeout(() => reject("Test timed out"), 5000);
+ });
+
+ video.srcObject = stream;
+ await video.play();
+
+ testImage();
}, "Basic video exchange");
+
+async function getInboundRTPStatsNumberOfDecodedFrames(connection)
+{
+ var report = await connection.getStats();
+ var framesDecoded;
+ report.forEach((statItem) => {
+ if (statItem.type === "inbound-rtp")
+ framesDecoded = statItem.framesDecoded;
+ });
+ return framesDecoded;
+}
+
+async function testFrameDecodedIncreased(connection, count, previousFramesNumber)
+{
+ if (previousFramesNumber === undefined) {
+ let number = await getInboundRTPStatsNumberOfDecodedFrames(connection);
+ await waitFor(1000);
+ return testFrameDecodedIncreased(connection, 0, number);
+ }
+
+ var number = await getInboundRTPStatsNumberOfDecodedFrames(connection);
+ if (previousFramesNumber && number > previousFramesNumber)
+ return;
+
+ if (count >= 20)
+ return Promise.reject("test increasing frame encoded timed out");
+
+ await waitFor(1000);
+ return testFrameDecodedIncreased(connection, ++count, previousFramesNumber);
+}
+
+async function testFrameDecodedDidNotIncreased(connection, count, previousFramesNumber)
+{
+ if (previousFramesNumber === undefined) {
+ let number = await getInboundRTPStatsNumberOfDecodedFrames(connection);
+ await waitFor(100);
+ return testFrameDecodedDidNotIncreased(connection, 0, number);
+ }
+
+ var number = await getInboundRTPStatsNumberOfDecodedFrames(connection);
+ if (previousFramesNumber && number == previousFramesNumber)
+ return;
+
+ if (count >= 20)
+ return Promise.reject("test increasing frame encoded timed out");
+
+ await waitFor(100);
+ return testFrameDecodedIncreased(connection, ++count, number);
+}
+
+promise_test(async (test) => {
+ let p = pc1.getSenders()[0].getParameters();
+ p.encodings[0].active = false;
+ await pc1.getSenders()[0].setParameters(p);
+
+ await testFrameDecodedDidNotIncreased(pc2);
+}, "Call setParameters to disable sending a given encoding");
+
+promise_test(async (test) => {
+ let p = pc1.getSenders()[0].getParameters();
+ p.encodings[0].active = true;
+ await pc1.getSenders()[0].setParameters(p);
+
+ await testFrameDecodedIncreased(pc2);
+}, "Call setParameters to reenable sending a given encoding");
</script>
</body>
</html>
Modified: trunk/Source/ThirdParty/libwebrtc/ChangeLog (239251 => 239252)
--- trunk/Source/ThirdParty/libwebrtc/ChangeLog 2018-12-15 10:03:19 UTC (rev 239251)
+++ trunk/Source/ThirdParty/libwebrtc/ChangeLog 2018-12-15 17:40:26 UTC (rev 239252)
@@ -1,3 +1,14 @@
+2018-12-15 Youenn Fablet <[email protected]>
+
+ Make RTCRtpSender.setParameters to activate specific encodings
+ https://bugs.webkit.org/show_bug.cgi?id=192732
+
+ Reviewed by Eric Carlson.
+
+ * Configurations/libwebrtc.iOS.exp:
+ * Configurations/libwebrtc.iOSsim.exp:
+ * Configurations/libwebrtc.mac.exp:
+
2018-12-14 Youenn Fablet <[email protected]>
kVTVideoEncoderSpecification_Usage should not be set if VCP is not enabled
Modified: trunk/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOS.exp (239251 => 239252)
--- trunk/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOS.exp 2018-12-15 10:03:19 UTC (rev 239251)
+++ trunk/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOS.exp 2018-12-15 17:40:26 UTC (rev 239252)
@@ -238,3 +238,11 @@
__ZTVN6webrtc20AudioSourceInterfaceE
__ZN6webrtc23PeerConnectionInterface21peer_connection_stateEv
__ZNK3rtc14NetworkManager16GetMdnsResponderEv
+__ZN6webrtc18RtpCodecParametersC1ERKS0_
+__ZN6webrtc18RtpCodecParametersC1Ev
+__ZN6webrtc18RtpCodecParametersD1Ev
+__ZN6webrtc13RtpParametersC1ERKS0_
+__ZN6webrtc16RtpFecParametersC1ERKS0_
+__ZN6webrtc16RtpFecParametersD1Ev
+__ZN6webrtc16RtpRtxParametersC1ERKS0_
+__ZN6webrtc16RtpRtxParametersD1Ev
Modified: trunk/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOSsim.exp (239251 => 239252)
--- trunk/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOSsim.exp 2018-12-15 10:03:19 UTC (rev 239251)
+++ trunk/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOSsim.exp 2018-12-15 17:40:26 UTC (rev 239252)
@@ -239,3 +239,11 @@
__ZTVN6webrtc20AudioSourceInterfaceE
__ZN6webrtc23PeerConnectionInterface21peer_connection_stateEv
__ZNK3rtc14NetworkManager16GetMdnsResponderEv
+__ZN6webrtc18RtpCodecParametersC1ERKS0_
+__ZN6webrtc18RtpCodecParametersC1Ev
+__ZN6webrtc18RtpCodecParametersD1Ev
+__ZN6webrtc13RtpParametersC1ERKS0_
+__ZN6webrtc16RtpFecParametersC1ERKS0_
+__ZN6webrtc16RtpFecParametersD1Ev
+__ZN6webrtc16RtpRtxParametersC1ERKS0_
+__ZN6webrtc16RtpRtxParametersD1Ev
Modified: trunk/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.mac.exp (239251 => 239252)
--- trunk/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.mac.exp 2018-12-15 10:03:19 UTC (rev 239251)
+++ trunk/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.mac.exp 2018-12-15 17:40:26 UTC (rev 239252)
@@ -239,3 +239,11 @@
__ZTVN6webrtc20AudioSourceInterfaceE
__ZN6webrtc23PeerConnectionInterface21peer_connection_stateEv
__ZNK3rtc14NetworkManager16GetMdnsResponderEv
+__ZN6webrtc18RtpCodecParametersC1ERKS0_
+__ZN6webrtc18RtpCodecParametersC1Ev
+__ZN6webrtc18RtpCodecParametersD1Ev
+__ZN6webrtc13RtpParametersC1ERKS0_
+__ZN6webrtc16RtpFecParametersC1ERKS0_
+__ZN6webrtc16RtpFecParametersD1Ev
+__ZN6webrtc16RtpRtxParametersC1ERKS0_
+__ZN6webrtc16RtpRtxParametersD1Ev
Modified: trunk/Source/WebCore/ChangeLog (239251 => 239252)
--- trunk/Source/WebCore/ChangeLog 2018-12-15 10:03:19 UTC (rev 239251)
+++ trunk/Source/WebCore/ChangeLog 2018-12-15 17:40:26 UTC (rev 239252)
@@ -1,3 +1,30 @@
+2018-12-15 Youenn Fablet <[email protected]>
+
+ Make RTCRtpSender.setParameters to activate specific encodings
+ https://bugs.webkit.org/show_bug.cgi?id=192732
+
+ Reviewed by Eric Carlson.
+
+ The conversion between libwebrtc and WebCore is lossy for send parameters.
+ Libwebrtc checking the differences of values, call to setParameters will often fail.
+
+ Given some parameters cannot be exposed, the sender backend keeps the
+ current set of parameters when gathered and reuses them when parameters are set.
+
+ For encodings, we only change activate/maxBitRate/maxFrameRate as
+ these are the most important parameters to be able to modify.
+
+ Covered by added tests in webrtc/video.html.
+
+ * Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.cpp:
+ (WebCore::LibWebRTCRtpSenderBackend::getParameters const):
+ (WebCore::LibWebRTCRtpSenderBackend::setParameters):
+ * Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.h:
+ * Modules/mediastream/libwebrtc/LibWebRTCUtils.cpp:
+ (WebCore::fromRTCRtpSendParameters):
+ (WebCore::fromRTCEncodingParameters): Deleted.
+ * Modules/mediastream/libwebrtc/LibWebRTCUtils.h:
+
2018-12-14 Joseph Pecoraro <[email protected]>
Web Inspector: Avoid creating and evaluating in the InspectorOverlay page on iOS as it is unused
Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.cpp (239251 => 239252)
--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.cpp 2018-12-15 10:03:19 UTC (rev 239251)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.cpp 2018-12-15 17:40:26 UTC (rev 239252)
@@ -117,7 +117,8 @@
if (!m_rtcSender)
return { };
- return toRTCRtpSendParameters(m_rtcSender->GetParameters());
+ m_currentParameters = m_rtcSender->GetParameters();
+ return toRTCRtpSendParameters(*m_currentParameters);
}
void LibWebRTCRtpSenderBackend::setParameters(const RTCRtpSendParameters& parameters, DOMPromiseDeferred<void>&& promise)
@@ -126,7 +127,17 @@
promise.reject(NotSupportedError);
return;
}
- auto error = m_rtcSender->SetParameters(fromRTCRtpSendParameters(parameters));
+
+ if (!m_currentParameters) {
+ promise.reject(Exception { InvalidStateError, "getParameters must be called before setParameters"_s });
+ return;
+ }
+
+ auto rtcParameters = WTFMove(*m_currentParameters);
+ updateRTCRtpSendParameters(parameters, rtcParameters);
+ m_currentParameters = std::nullopt;
+
+ auto error = m_rtcSender->SetParameters(rtcParameters);
if (!error.ok()) {
promise.reject(Exception { InvalidStateError, error.message() });
return;
Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.h (239251 => 239252)
--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.h 2018-12-15 10:03:19 UTC (rev 239251)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.h 2018-12-15 17:40:26 UTC (rev 239252)
@@ -114,6 +114,7 @@
WeakPtr<LibWebRTCPeerConnectionBackend> m_peerConnectionBackend;
rtc::scoped_refptr<webrtc::RtpSenderInterface> m_rtcSender;
Source m_source;
+ mutable std::optional<webrtc::RtpParameters> m_currentParameters;
};
} // namespace WebCore
Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCUtils.cpp (239251 => 239252)
--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCUtils.cpp 2018-12-15 10:03:19 UTC (rev 239251)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCUtils.cpp 2018-12-15 17:40:26 UTC (rev 239252)
@@ -72,31 +72,6 @@
return parameters;
}
-static inline webrtc::RtpEncodingParameters fromRTCEncodingParameters(const RTCRtpEncodingParameters& parameters)
-{
- webrtc::RtpEncodingParameters rtcParameters;
-
- if (parameters.dtx) {
- switch (*parameters.dtx) {
- case RTCDtxStatus::Disabled:
- rtcParameters.dtx = webrtc::DtxStatus::DISABLED;
- break;
- case RTCDtxStatus::Enabled:
- rtcParameters.dtx = webrtc::DtxStatus::ENABLED;
- }
- }
- rtcParameters.active = parameters.active;
- if (parameters.maxBitrate)
- rtcParameters.max_bitrate_bps = parameters.maxBitrate;
- if (parameters.maxFramerate)
- rtcParameters.max_framerate = parameters.maxFramerate;
- rtcParameters.rid = parameters.rid.utf8().data();
- if (parameters.scaleResolutionDownBy != 1)
- rtcParameters.scale_resolution_down_by = parameters.scaleResolutionDownBy;
-
- return rtcParameters;
-}
-
static inline RTCRtpHeaderExtensionParameters toRTCHeaderExtensionParameters(const webrtc::RtpHeaderExtensionParameters& rtcParameters)
{
RTCRtpHeaderExtensionParameters parameters;
@@ -185,13 +160,28 @@
return parameters;
}
-webrtc::RtpParameters fromRTCRtpSendParameters(const RTCRtpSendParameters& parameters)
+void updateRTCRtpSendParameters(const RTCRtpSendParameters& parameters, webrtc::RtpParameters& currentParameters)
{
- webrtc::RtpParameters rtcParameters;
+ webrtc::RtpParameters rtcParameters = currentParameters;
+
rtcParameters.transaction_id = parameters.transactionId.utf8().data();
- for (auto& encoding : parameters.encodings)
- rtcParameters.encodings.push_back(fromRTCEncodingParameters(encoding));
+ if (parameters.encodings.size() != rtcParameters.encodings.size()) {
+ // If encodings size is different, setting parameters will fail. Let's make it so.
+ rtcParameters.encodings.clear();
+ return;
+ }
+
+ // We copy all current encodings parameters and only update parameters that can actually be usefully updated.
+ for (size_t i = 0; i < parameters.encodings.size(); ++i) {
+ rtcParameters.encodings[i].active = parameters.encodings[i].active;
+ if (parameters.encodings[i].maxBitrate)
+ rtcParameters.encodings[i].max_bitrate_bps = parameters.encodings[i].maxBitrate;
+ if (parameters.encodings[i].maxFramerate)
+ rtcParameters.encodings[i].max_framerate = parameters.encodings[i].maxFramerate;
+ }
+
+ rtcParameters.header_extensions.clear();
for (auto& extension : parameters.headerExtensions)
rtcParameters.header_extensions.push_back(fromRTCHeaderExtensionParameters(extension));
// Codecs parameters are readonly
@@ -207,7 +197,6 @@
rtcParameters.degradation_preference = webrtc::DegradationPreference::BALANCED;
break;
}
- return rtcParameters;
}
RTCRtpTransceiverDirection toRTCRtpTransceiverDirection(webrtc::RtpTransceiverDirection rtcDirection)
Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCUtils.h (239251 => 239252)
--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCUtils.h 2018-12-15 10:03:19 UTC (rev 239251)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCUtils.h 2018-12-15 17:40:26 UTC (rev 239252)
@@ -44,9 +44,9 @@
enum class RTCRtpTransceiverDirection;
RTCRtpParameters toRTCRtpParameters(const webrtc::RtpParameters&);
-webrtc::RtpParameters fromRTCRtpParameters(const RTCRtpParameters&);
+void updateRTCRtpSendParameters(const RTCRtpSendParameters&, webrtc::RtpParameters&);
RTCRtpSendParameters toRTCRtpSendParameters(const webrtc::RtpParameters&);
-webrtc::RtpParameters fromRTCRtpSendParameters(const RTCRtpSendParameters&);
+webrtc::RtpParameters fromRTCRtpSendParameters(const RTCRtpSendParameters&, const webrtc::RtpParameters& currentParameters);
RTCRtpTransceiverDirection toRTCRtpTransceiverDirection(webrtc::RtpTransceiverDirection);
webrtc::RtpTransceiverDirection fromRTCRtpTransceiverDirection(RTCRtpTransceiverDirection);
