https://bugs.webkit.org/show_bug.cgi?id=192732
Reviewed by Eric Carlson.
Source/ThirdParty/libwebrtc:
* Configurations/libwebrtc.iOS.exp:
* Configurations/libwebrtc.iOSsim.exp:
* Configurations/libwebrtc.mac.exp:
Source/WebCore:
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:
LayoutTests:
* webrtc/video-expected.txt:
* webrtc/video.html:
git-svn-id: https://svn.webkit.org/repository/webkit/trunk@239252
268f45cc-cd09-0410-ab3c-
d52691b4dbfc
+2018-12-15 Youenn Fablet <youenn@apple.com>
+
+ 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 <nvasilyev@apple.com>
Web Inspector: Styles: toggling selected properties may cause data corruption
PASS Basic video exchange
+PASS Call setParameters to disable sending a given encoding
+PASS Call setParameters to reenable sending a given encoding
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>
+2018-12-15 Youenn Fablet <youenn@apple.com>
+
+ 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 <youenn@apple.com>
kVTVideoEncoderSpecification_Usage should not be set if VCP is not enabled
__ZTVN6webrtc20AudioSourceInterfaceE
__ZN6webrtc23PeerConnectionInterface21peer_connection_stateEv
__ZNK3rtc14NetworkManager16GetMdnsResponderEv
+__ZN6webrtc18RtpCodecParametersC1ERKS0_
+__ZN6webrtc18RtpCodecParametersC1Ev
+__ZN6webrtc18RtpCodecParametersD1Ev
+__ZN6webrtc13RtpParametersC1ERKS0_
+__ZN6webrtc16RtpFecParametersC1ERKS0_
+__ZN6webrtc16RtpFecParametersD1Ev
+__ZN6webrtc16RtpRtxParametersC1ERKS0_
+__ZN6webrtc16RtpRtxParametersD1Ev
__ZTVN6webrtc20AudioSourceInterfaceE
__ZN6webrtc23PeerConnectionInterface21peer_connection_stateEv
__ZNK3rtc14NetworkManager16GetMdnsResponderEv
+__ZN6webrtc18RtpCodecParametersC1ERKS0_
+__ZN6webrtc18RtpCodecParametersC1Ev
+__ZN6webrtc18RtpCodecParametersD1Ev
+__ZN6webrtc13RtpParametersC1ERKS0_
+__ZN6webrtc16RtpFecParametersC1ERKS0_
+__ZN6webrtc16RtpFecParametersD1Ev
+__ZN6webrtc16RtpRtxParametersC1ERKS0_
+__ZN6webrtc16RtpRtxParametersD1Ev
__ZTVN6webrtc20AudioSourceInterfaceE
__ZN6webrtc23PeerConnectionInterface21peer_connection_stateEv
__ZNK3rtc14NetworkManager16GetMdnsResponderEv
+__ZN6webrtc18RtpCodecParametersC1ERKS0_
+__ZN6webrtc18RtpCodecParametersC1Ev
+__ZN6webrtc18RtpCodecParametersD1Ev
+__ZN6webrtc13RtpParametersC1ERKS0_
+__ZN6webrtc16RtpFecParametersC1ERKS0_
+__ZN6webrtc16RtpFecParametersD1Ev
+__ZN6webrtc16RtpRtxParametersC1ERKS0_
+__ZN6webrtc16RtpRtxParametersD1Ev
+2018-12-15 Youenn Fablet <youenn@apple.com>
+
+ 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 <pecoraro@apple.com>
Web Inspector: Avoid creating and evaluating in the InspectorOverlay page on iOS as it is unused
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)
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;
WeakPtr<LibWebRTCPeerConnectionBackend> m_peerConnectionBackend;
rtc::scoped_refptr<webrtc::RtpSenderInterface> m_rtcSender;
Source m_source;
+ mutable std::optional<webrtc::RtpParameters> m_currentParameters;
};
} // namespace WebCore
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;
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
rtcParameters.degradation_preference = webrtc::DegradationPreference::BALANCED;
break;
}
- return rtcParameters;
}
RTCRtpTransceiverDirection toRTCRtpTransceiverDirection(webrtc::RtpTransceiverDirection rtcDirection)
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);
git://git.webkit.org / WebKit-https.git / commitdiff