Attachment 348946 Details for Bug 189307 – Patch

[patch] Patch

bug-189307-20180905111233.patch (text/plain), 24.54 KB, created by youenn fablet on 2018-09-05 11:12:34 PDT

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Creator: youenn fablet

Created: 2018-09-05 11:12:34 PDT

Size: 24.54 KB

patch

obsolete

>Subversion Revision: 235669
>diff --git a/Source/ThirdParty/libwebrtc/ChangeLog b/Source/ThirdParty/libwebrtc/ChangeLog
>index 1865dfaf0ac140ac2f0f6c401188f22d58938e96..3ab6f40ed11ee64ef877e08d83de4e5b04455876 100644
>--- a/Source/ThirdParty/libwebrtc/ChangeLog
>+++ b/Source/ThirdParty/libwebrtc/ChangeLog
>@@ -1,3 +1,14 @@
>+2018-09-05  Youenn Fablet  <youenn@apple.com>
>+
>+        Expose RTCRtpSender.setParameters
>+        https://bugs.webkit.org/show_bug.cgi?id=189307
>+
>+        Reviewed by NOBODY (OOPS!).
>+
>+        * Configurations/libwebrtc.iOS.exp:
>+        * Configurations/libwebrtc.iOSsim.exp:
>+        * Configurations/libwebrtc.mac.exp:
>+
> 2018-08-29  David Kilzer  <ddkilzer@apple.com>
> 
>         Remove empty directories from from svn.webkit.org repository
>diff --git a/Source/WebCore/ChangeLog b/Source/WebCore/ChangeLog
>index 476db707225808f0732d9ac2ecbf829f38096907..2b7fb699b4cfb5e8ccacda5b9f001438d167859c 100644
>--- a/Source/WebCore/ChangeLog
>+++ b/Source/WebCore/ChangeLog
>@@ -1,3 +1,41 @@
>+2018-09-05  Youenn Fablet  <youenn@apple.com>
>+
>+        Expose RTCRtpSender.setParameters
>+        https://bugs.webkit.org/show_bug.cgi?id=189307
>+
>+        Reviewed by NOBODY (OOPS!).
>+
>+        Implement RTCRtpSender.setParameters.
>+        This is an incomplete implementation as we need to refresh parameters as per the spec and testing is much easier with full transceiver support.
>+        Implementation uses sender backend to set the parameters at libwebrtc level.
>+        Fix the case of a stopped sender/transceiver in which case promise should be rejected.
>+        Covered by rebased and modified tests.
>+
>+        * Modules/mediastream/RTCRtpParameters.h:
>+        * Modules/mediastream/RTCRtpSender.cpp:
>+        (WebCore::RTCRtpSender::setParameters):
>+        * Modules/mediastream/RTCRtpSender.h:
>+        * Modules/mediastream/RTCRtpSender.idl:
>+        * Modules/mediastream/RTCRtpSenderBackend.h:
>+        * Modules/mediastream/RTCRtpTransceiver.cpp:
>+        (WebCore::RTCRtpTransceiver::stop):
>+        * Modules/mediastream/RTCRtpTransceiver.h:
>+        * Modules/mediastream/libwebrtc/LibWebRTCRtpReceiverBackend.cpp:
>+        (WebCore::LibWebRTCRtpReceiverBackend::getParameters):
>+        * Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.cpp:
>+        (WebCore::LibWebRTCRtpSenderBackend::getParameters const):
>+        (WebCore::LibWebRTCRtpSenderBackend::setParameters):
>+        * Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.h:
>+        * Modules/mediastream/libwebrtc/LibWebRTCUtils.cpp:
>+        (WebCore::toRTCEncodingParameters):
>+        (WebCore::fromRTCEncodingParameters):
>+        (WebCore::toRTCHeaderExtensionParameters):
>+        (WebCore::fromRTCHeaderExtensionParameters):
>+        (WebCore::toRTCCodecParameters):
>+        (WebCore::toRTCRtpParameters):
>+        (WebCore::fromRTCRtpParameters):
>+        * Modules/mediastream/libwebrtc/LibWebRTCUtils.h:
>+
> 2018-09-05  Youenn Fablet  <youenn@apple.com>
> 
>         Move replaceTrack logic to LibWebRTCPeerConnectionBackend
>diff --git a/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOS.exp b/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOS.exp
>index 8478b2841526d4cce6169a76de2a04565d150287..d64cdf08b41b39a57afdb13fb8a78716a650a52d 100644
>--- a/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOS.exp
>+++ b/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOS.exp
>@@ -206,3 +206,9 @@ __ZN3rtc13PacketOptionsD1Ev
> __ZN3rtc19PacketSocketFactory21CreateClientTcpSocketERKNS_13SocketAddressES3_RKNS_9ProxyInfoERKNSt3__112basic_stringIcNS7_11char_traitsIcEENS7_9allocatorIcEEEERKNS_22PacketSocketTcpOptionsE
> __ZN3rtc13PacketOptionsC1Ev
> __ZN6webrtc27CreatePeerConnectionFactoryEPN3rtc6ThreadES2_S2_NS0_13scoped_refptrINS_17AudioDeviceModuleEEENS3_INS_19AudioEncoderFactoryEEENS3_INS_19AudioDecoderFactoryEEENSt3__110unique_ptrINS_19VideoEncoderFactoryENSA_14default_deleteISC_EEEENSB_INS_19VideoDecoderFactoryENSD_ISG_EEEENS3_INS_10AudioMixerEEENS3_INS_15AudioProcessingEEE
>+__ZNK6webrtc8RTCError7messageEv
>+__ZN6webrtc12RtpExtensionC1Ev
>+__ZN6webrtc12RtpExtensionD1Ev
>+__ZN6webrtc21RtpEncodingParametersC1ERKS0_
>+__ZN6webrtc21RtpEncodingParametersC1Ev
>+__ZN6webrtc21RtpEncodingParametersD1Ev
>diff --git a/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOSsim.exp b/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOSsim.exp
>index 8478b2841526d4cce6169a76de2a04565d150287..4fc1c4abd7a524b70ce8d50a333c180c63f6d4ff 100644
>--- a/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOSsim.exp
>+++ b/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOSsim.exp
>@@ -206,3 +206,10 @@ __ZN3rtc13PacketOptionsD1Ev
> __ZN3rtc19PacketSocketFactory21CreateClientTcpSocketERKNS_13SocketAddressES3_RKNS_9ProxyInfoERKNSt3__112basic_stringIcNS7_11char_traitsIcEENS7_9allocatorIcEEEERKNS_22PacketSocketTcpOptionsE
> __ZN3rtc13PacketOptionsC1Ev
> __ZN6webrtc27CreatePeerConnectionFactoryEPN3rtc6ThreadES2_S2_NS0_13scoped_refptrINS_17AudioDeviceModuleEEENS3_INS_19AudioEncoderFactoryEEENS3_INS_19AudioDecoderFactoryEEENSt3__110unique_ptrINS_19VideoEncoderFactoryENSA_14default_deleteISC_EEEENSB_INS_19VideoDecoderFactoryENSD_ISG_EEEENS3_INS_10AudioMixerEEENS3_INS_15AudioProcessingEEE
>+__ZNK6webrtc8RTCError7messageEv
>+__ZNK6webrtc8RTCError7messageEv
>+__ZN6webrtc12RtpExtensionC1Ev
>+__ZN6webrtc12RtpExtensionD1Ev
>+__ZN6webrtc21RtpEncodingParametersC1ERKS0_
>+__ZN6webrtc21RtpEncodingParametersC1Ev
>+__ZN6webrtc21RtpEncodingParametersD1Ev
>diff --git a/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.mac.exp b/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.mac.exp
>index 8478b2841526d4cce6169a76de2a04565d150287..4fc1c4abd7a524b70ce8d50a333c180c63f6d4ff 100644
>--- a/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.mac.exp
>+++ b/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.mac.exp
>@@ -206,3 +206,10 @@ __ZN3rtc13PacketOptionsD1Ev
> __ZN3rtc19PacketSocketFactory21CreateClientTcpSocketERKNS_13SocketAddressES3_RKNS_9ProxyInfoERKNSt3__112basic_stringIcNS7_11char_traitsIcEENS7_9allocatorIcEEEERKNS_22PacketSocketTcpOptionsE
> __ZN3rtc13PacketOptionsC1Ev
> __ZN6webrtc27CreatePeerConnectionFactoryEPN3rtc6ThreadES2_S2_NS0_13scoped_refptrINS_17AudioDeviceModuleEEENS3_INS_19AudioEncoderFactoryEEENS3_INS_19AudioDecoderFactoryEEENSt3__110unique_ptrINS_19VideoEncoderFactoryENSA_14default_deleteISC_EEEENSB_INS_19VideoDecoderFactoryENSD_ISG_EEEENS3_INS_10AudioMixerEEENS3_INS_15AudioProcessingEEE
>+__ZNK6webrtc8RTCError7messageEv
>+__ZNK6webrtc8RTCError7messageEv
>+__ZN6webrtc12RtpExtensionC1Ev
>+__ZN6webrtc12RtpExtensionD1Ev
>+__ZN6webrtc21RtpEncodingParametersC1ERKS0_
>+__ZN6webrtc21RtpEncodingParametersC1Ev
>+__ZN6webrtc21RtpEncodingParametersD1Ev
>diff --git a/Source/WebCore/Modules/mediastream/RTCRtpParameters.h b/Source/WebCore/Modules/mediastream/RTCRtpParameters.h
>index f0fc8eeee80766b85bcb2377b2fbca23af65a75d..20171b5e8a2f768c7551bfed767b64186589080c 100644
>--- a/Source/WebCore/Modules/mediastream/RTCRtpParameters.h
>+++ b/Source/WebCore/Modules/mediastream/RTCRtpParameters.h
>@@ -57,7 +57,7 @@ struct RTCRtpParameters {
>         unsigned long ssrc { 0 };
>         RtxParameters rtx;
>         FecParameters fec;
>-        DtxStatus dtx { DtxStatus::Disabled };
>+        std::optional<DtxStatus> dtx;
>         bool active { false};
>         PriorityType priority { PriorityType::Medium };
>         unsigned long maxBitrate { 0 };
>diff --git a/Source/WebCore/Modules/mediastream/RTCRtpSender.cpp b/Source/WebCore/Modules/mediastream/RTCRtpSender.cpp
>index 5acdd466f4934049c3b497f03fccd7966647bfe1..2b5b3c605926f73a7fc26e2a17b00672f23fbd3a 100644
>--- a/Source/WebCore/Modules/mediastream/RTCRtpSender.cpp
>+++ b/Source/WebCore/Modules/mediastream/RTCRtpSender.cpp
>@@ -99,6 +99,15 @@ RTCRtpParameters RTCRtpSender::getParameters()
>     return m_backend->getParameters();
> }
> 
>+void RTCRtpSender::setParameters(const RTCRtpParameters& parameters, DOMPromiseDeferred<void>&& promise)
>+{
>+    if (isStopped()) {
>+        promise.reject(InvalidStateError);
>+        return;
>+    }
>+    return m_backend->setParameters(parameters, WTFMove(promise));
>+}
>+
> } // namespace WebCore
> 
> #endif // ENABLE(WEB_RTC)
>diff --git a/Source/WebCore/Modules/mediastream/RTCRtpSender.h b/Source/WebCore/Modules/mediastream/RTCRtpSender.h
>index a5d30924e3a03ba741eef8985e0e7503bbbd9a91..45b89014461b10be3b0b2183239eca7a9aa041ee 100644
>--- a/Source/WebCore/Modules/mediastream/RTCRtpSender.h
>+++ b/Source/WebCore/Modules/mediastream/RTCRtpSender.h
>@@ -57,6 +57,8 @@ public:
>     void replaceTrack(RefPtr<MediaStreamTrack>&&, DOMPromiseDeferred<void>&&);
> 
>     RTCRtpParameters getParameters();
>+    void setParameters(const RTCRtpParameters&, DOMPromiseDeferred<void>&&);
>+
>     RTCRtpSenderBackend* backend() { return m_backend.get(); }
> 
> private:
>diff --git a/Source/WebCore/Modules/mediastream/RTCRtpSender.idl b/Source/WebCore/Modules/mediastream/RTCRtpSender.idl
>index 3f3fe419a66d558ed1b4d851a63d0e3a49e54e72..c8dcd97ed2708b180493793036b6d48a3bd285e3 100644
>--- a/Source/WebCore/Modules/mediastream/RTCRtpSender.idl
>+++ b/Source/WebCore/Modules/mediastream/RTCRtpSender.idl
>@@ -37,7 +37,8 @@
>     // FIXME 169662: missing transport
>     // FIXME 169662: missing rtcpTransport
>     // FIXME 169662: missing getCapabilities
>-    // FIXME 169662: missing setParameters
>+    // FIXME 169662: missing setStreams
>     RTCRtpParameters getParameters();
>+    Promise<void> setParameters(RTCRtpParameters parameters);
>     Promise<void> replaceTrack(MediaStreamTrack? withTrack);
> };
>diff --git a/Source/WebCore/Modules/mediastream/RTCRtpSenderBackend.h b/Source/WebCore/Modules/mediastream/RTCRtpSenderBackend.h
>index 476543c98c6fa6ca465015c1bc58fcee7a97a431..4ad09ecaf43b04132599ffb1ff82bd16324206df 100644
>--- a/Source/WebCore/Modules/mediastream/RTCRtpSenderBackend.h
>+++ b/Source/WebCore/Modules/mediastream/RTCRtpSenderBackend.h
>@@ -38,6 +38,7 @@ class RTCRtpSenderBackend {
> public:
>     virtual void replaceTrack(RTCRtpSender&, RefPtr<MediaStreamTrack>&&, DOMPromiseDeferred<void>&&) = 0;
>     virtual RTCRtpParameters getParameters() const = 0;
>+    virtual void setParameters(const RTCRtpParameters&, DOMPromiseDeferred<void>&&) = 0;
>     virtual ~RTCRtpSenderBackend() = default;
> };
> 
>diff --git a/Source/WebCore/Modules/mediastream/RTCRtpTransceiver.cpp b/Source/WebCore/Modules/mediastream/RTCRtpTransceiver.cpp
>index 848c3df5d0e811cfb50742818800835290e07b19..ad4c171667eacb0318de44377223fa08a5a4a4f2 100644
>--- a/Source/WebCore/Modules/mediastream/RTCRtpTransceiver.cpp
>+++ b/Source/WebCore/Modules/mediastream/RTCRtpTransceiver.cpp
>@@ -102,6 +102,13 @@ void RTCRtpTransceiver::disableSendingDirection()
>         m_direction = RTCRtpTransceiverDirection::Inactive;
> }
> 
>+void RTCRtpTransceiver::stop()
>+{
>+    m_stopped = true;
>+    m_receiver->stop();
>+    m_sender->stop();
>+}
>+
> void RtpTransceiverSet::append(Ref<RTCRtpTransceiver>&& transceiver)
> {
>     m_senders.append(transceiver->sender());
>diff --git a/Source/WebCore/Modules/mediastream/RTCRtpTransceiver.h b/Source/WebCore/Modules/mediastream/RTCRtpTransceiver.h
>index 1a1cac92aa848a5431d5a98286cc46fb0732e607..3e71be7933e44b0624c2a161cc5eb65e56f631d2 100644
>--- a/Source/WebCore/Modules/mediastream/RTCRtpTransceiver.h
>+++ b/Source/WebCore/Modules/mediastream/RTCRtpTransceiver.h
>@@ -68,7 +68,7 @@ public:
>     RTCRtpReceiver& receiver() { return m_receiver.get(); }
> 
>     bool stopped() const { return m_stopped; }
>-    void stop() { m_stopped = true; }
>+    void stop();
> 
>     // FIXME: Temporary solution to keep track of ICE states for this transceiver. Later, each
>     // sender and receiver will have up to two DTLS transports, which in turn will have an ICE
>diff --git a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpReceiverBackend.cpp b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpReceiverBackend.cpp
>index 4df4e7ccafe63133b835b01db2d0f7baeaa09405..5406d864fda0493c860884aa7fb315c29ddc6667 100644
>--- a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpReceiverBackend.cpp
>+++ b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpReceiverBackend.cpp
>@@ -33,7 +33,7 @@ namespace WebCore {
> 
> RTCRtpParameters LibWebRTCRtpReceiverBackend::getParameters()
> {
>-    return fillRtpParameters(m_rtcReceiver->GetParameters());
>+    return toRTCRtpParameters(m_rtcReceiver->GetParameters());
> }
> 
> } // namespace WebCore
>diff --git a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.cpp b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.cpp
>index 51a78ae31dbaff5bb5f9ec5b1125b7af9d012c37..9371820034757a6793df3a66a6bf3540d2d8b257 100644
>--- a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.cpp
>+++ b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.cpp
>@@ -49,7 +49,21 @@ RTCRtpParameters LibWebRTCRtpSenderBackend::getParameters() const
>     if (!m_rtcSender)
>         return { };
> 
>-    return fillRtpParameters(m_rtcSender->GetParameters());
>+    return toRTCRtpParameters(m_rtcSender->GetParameters());
>+}
>+
>+void LibWebRTCRtpSenderBackend::setParameters(const RTCRtpParameters& parameters, DOMPromiseDeferred<void>&& promise)
>+{
>+    if (!m_rtcSender) {
>+        promise.reject(NotSupportedError);
>+        return;
>+    }
>+    auto error = m_rtcSender->SetParameters(fromRTCRtpParameters(parameters));
>+    if (!error.ok()) {
>+        promise.reject(Exception { InvalidStateError, error.message() });
>+        return;
>+    }
>+    promise.resolve();
> }
> 
> } // namespace WebCore
>diff --git a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.h b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.h
>index 686fa88f3ed516b0523b2319f01a6f9d85cccd68..66927f96451d51e1640160768f6a7fdb3af2ab0f 100644
>--- a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.h
>+++ b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.h
>@@ -94,6 +94,7 @@ public:
> private:
>     void replaceTrack(RTCRtpSender&, RefPtr<MediaStreamTrack>&&, DOMPromiseDeferred<void>&&) final;
>     RTCRtpParameters getParameters() const final;
>+    void setParameters(const RTCRtpParameters&, DOMPromiseDeferred<void>&&) final;
> 
>     WeakPtr<LibWebRTCPeerConnectionBackend> m_peerConnectionBackend;
>     rtc::scoped_refptr<webrtc::RtpSenderInterface> m_rtcSender;
>diff --git a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCUtils.cpp b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCUtils.cpp
>index 4e8541158952a869223ddc7525326147181978e3..2f3c10ce05985f71355f614556116552d6f29939 100644
>--- a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCUtils.cpp
>+++ b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCUtils.cpp
>@@ -34,7 +34,7 @@
> 
> namespace WebCore {
> 
>-static inline RTCRtpParameters::EncodingParameters fillEncodingParameters(const webrtc::RtpEncodingParameters& rtcParameters)
>+static inline RTCRtpParameters::EncodingParameters toRTCEncodingParameters(const webrtc::RtpEncodingParameters& rtcParameters)
> {
>     RTCRtpParameters::EncodingParameters parameters;
> 
>@@ -65,7 +65,32 @@ static inline RTCRtpParameters::EncodingParameters fillEncodingParameters(const
>     return parameters;
> }
> 
>-static inline RTCRtpParameters::HeaderExtensionParameters fillHeaderExtensionParameters(const webrtc::RtpHeaderExtensionParameters& rtcParameters)
>+static inline webrtc::RtpEncodingParameters fromRTCEncodingParameters(const RTCRtpParameters::EncodingParameters& parameters)
>+{
>+    webrtc::RtpEncodingParameters rtcParameters;
>+
>+    if (parameters.dtx) {
>+        switch (*parameters.dtx) {
>+        case RTCRtpParameters::DtxStatus::Disabled:
>+            rtcParameters.dtx = webrtc::DtxStatus::DISABLED;
>+            break;
>+        case RTCRtpParameters::DtxStatus::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 RTCRtpParameters::HeaderExtensionParameters toRTCHeaderExtensionParameters(const webrtc::RtpHeaderExtensionParameters& rtcParameters)
> {
>     RTCRtpParameters::HeaderExtensionParameters parameters;
> 
>@@ -75,7 +100,17 @@ static inline RTCRtpParameters::HeaderExtensionParameters fillHeaderExtensionPar
>     return parameters;
> }
> 
>-static inline RTCRtpParameters::CodecParameters fillCodecParameters(const webrtc::RtpCodecParameters& rtcParameters)
>+static inline webrtc::RtpHeaderExtensionParameters fromRTCHeaderExtensionParameters(const RTCRtpParameters::HeaderExtensionParameters& parameters)
>+{
>+    webrtc::RtpHeaderExtensionParameters rtcParameters;
>+
>+    rtcParameters.uri = parameters.uri.utf8().data();
>+    rtcParameters.id = parameters.id;
>+
>+    return rtcParameters;
>+}
>+
>+static inline RTCRtpParameters::CodecParameters toRTCCodecParameters(const webrtc::RtpCodecParameters& rtcParameters)
> {
>     RTCRtpParameters::CodecParameters parameters;
> 
>@@ -89,17 +124,17 @@ static inline RTCRtpParameters::CodecParameters fillCodecParameters(const webrtc
>     return parameters;
> }
> 
>-RTCRtpParameters fillRtpParameters(const webrtc::RtpParameters& rtcParameters)
>+RTCRtpParameters toRTCRtpParameters(const webrtc::RtpParameters& rtcParameters)
> {
>     RTCRtpParameters parameters;
> 
>     parameters.transactionId = fromStdString(rtcParameters.transaction_id);
>     for (auto& rtcEncoding : rtcParameters.encodings)
>-        parameters.encodings.append(fillEncodingParameters(rtcEncoding));
>+        parameters.encodings.append(toRTCEncodingParameters(rtcEncoding));
>     for (auto& extension : rtcParameters.header_extensions)
>-        parameters.headerExtensions.append(fillHeaderExtensionParameters(extension));
>+        parameters.headerExtensions.append(toRTCHeaderExtensionParameters(extension));
>     for (auto& codec : rtcParameters.codecs)
>-        parameters.codecs.append(fillCodecParameters(codec));
>+        parameters.codecs.append(toRTCCodecParameters(codec));
> 
>     switch (rtcParameters.degradation_preference) {
>     // FIXME: Support DegradationPreference::DISABLED.
>@@ -117,6 +152,31 @@ RTCRtpParameters fillRtpParameters(const webrtc::RtpParameters& rtcParameters)
>     return parameters;
> }
> 
>+webrtc::RtpParameters fromRTCRtpParameters(const RTCRtpParameters& parameters)
>+{
>+    webrtc::RtpParameters rtcParameters;
>+    rtcParameters.transaction_id = parameters.transactionId.utf8().data();
>+
>+    for (auto& encoding : parameters.encodings)
>+        rtcParameters.encodings.push_back(fromRTCEncodingParameters(encoding));
>+    for (auto& extension : parameters.headerExtensions)
>+        rtcParameters.header_extensions.push_back(fromRTCHeaderExtensionParameters(extension));
>+    // Codecs parameters are readonly
>+
>+    switch (parameters.degradationPreference) {
>+    case RTCRtpParameters::DegradationPreference::MaintainFramerate:
>+        rtcParameters.degradation_preference = webrtc::DegradationPreference::MAINTAIN_FRAMERATE;
>+        break;
>+    case RTCRtpParameters::DegradationPreference::MaintainResolution:
>+        rtcParameters.degradation_preference = webrtc::DegradationPreference::MAINTAIN_RESOLUTION;
>+        break;
>+    case RTCRtpParameters::DegradationPreference::Balanced:
>+        rtcParameters.degradation_preference = webrtc::DegradationPreference::BALANCED;
>+        break;
>+    }
>+    return rtcParameters;
>+}
>+
> }; // namespace WebCore
> 
> #endif // USE(LIBWEBRTC)
>diff --git a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCUtils.h b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCUtils.h
>index f26c3cf35e95dc38a1d78aa8cc626cb4c92ee012..ba63751f1a24deeb1569c323cadcb9bc42e9dfae 100644
>--- a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCUtils.h
>+++ b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCUtils.h
>@@ -36,7 +36,8 @@ namespace WebCore {
> 
> struct RTCRtpParameters;
> 
>-RTCRtpParameters fillRtpParameters(const webrtc::RtpParameters&);
>+RTCRtpParameters toRTCRtpParameters(const webrtc::RtpParameters&);
>+webrtc::RtpParameters fromRTCRtpParameters(const RTCRtpParameters&);
> 
> inline String fromStdString(const std::string& value)
> {
>diff --git a/LayoutTests/ChangeLog b/LayoutTests/ChangeLog
>index 3879ee7cb12346e495da0131fdcac40fc100256f..b9a5c045741bc089ffc546919fbe62833f82efef 100644
>--- a/LayoutTests/ChangeLog
>+++ b/LayoutTests/ChangeLog
>@@ -1,3 +1,12 @@
>+2018-09-05  Youenn Fablet  <youenn@apple.com>
>+
>+        Expose RTCRtpSender.setParameters
>+        https://bugs.webkit.org/show_bug.cgi?id=189307
>+
>+        Reviewed by NOBODY (OOPS!).
>+
>+        * webrtc/video-getParameters.html:
>+
> 2018-09-05  Ms2ger  <Ms2ger@igalia.com>
> 
>         GTK-focused gardening
>diff --git a/LayoutTests/imported/w3c/ChangeLog b/LayoutTests/imported/w3c/ChangeLog
>index cf7b31b0ff0b0823f5dc2948d8d5e026eb1b9ac9..2a84347bd0834b9277dbdcfd354e6d2d79b6429a 100644
>--- a/LayoutTests/imported/w3c/ChangeLog
>+++ b/LayoutTests/imported/w3c/ChangeLog
>@@ -1,3 +1,13 @@
>+2018-09-05  Youenn Fablet  <youenn@apple.com>
>+
>+        Expose RTCRtpSender.setParameters
>+        https://bugs.webkit.org/show_bug.cgi?id=189307
>+
>+        Reviewed by NOBODY (OOPS!).
>+
>+        * web-platform-tests/webrtc/RTCRtpSender-replaceTrack-expected.txt:
>+        * web-platform-tests/webrtc/RTCRtpSender-setParameters-expected.txt:
>+
> 2018-09-04  Frederic Wang  <fwang@igalia.com>
> 
>         Add basic support for ScrollIntoViewOptions
>diff --git a/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpSender-replaceTrack-expected.txt b/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpSender-replaceTrack-expected.txt
>index 71f717fe1fe6af4689b36d7b1bf7f5ca23c0a6c9..fc64b50113502989e5f2b5c3e52f6c8efdd5f3ec 100644
>--- a/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpSender-replaceTrack-expected.txt
>+++ b/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpSender-replaceTrack-expected.txt
>@@ -1,7 +1,7 @@
> 
> PASS Calling replaceTrack on closed connection should reject with InvalidStateError 
> PASS Calling replaceTrack with track of different kind should reject with TypeError 
>-FAIL Calling replaceTrack on stopped sender should reject with InvalidStateError assert_unreached: Should have rejected: undefined Reached unreachable code
>+PASS Calling replaceTrack on stopped sender should reject with InvalidStateError 
> PASS Calling replaceTrack on sender with null track and not set to session description should resolve with sender.track set to given track 
> PASS Calling replaceTrack on sender not set to session description should resolve with sender.track set to given track 
> PASS Calling replaceTrack(null) on sender not set to session description should resolve with sender.track set to null 
>diff --git a/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpSender-setParameters-expected.txt b/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpSender-setParameters-expected.txt
>index 156bd67afa343983641d4c7454ce743c5c30a88d..78a6005a9d7e0963d6ee786f0ad9fb8b8af013bc 100644
>--- a/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpSender-setParameters-expected.txt
>+++ b/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpSender-setParameters-expected.txt
>@@ -1,3 +1,3 @@
> 
>-FAIL setParameters() when transceiver is stopped should reject with InvalidStateError sender.setParameters is not a function. (In 'sender.setParameters(param)', 'sender.setParameters' is undefined)
>+PASS setParameters() when transceiver is stopped should reject with InvalidStateError 
> 
>diff --git a/LayoutTests/webrtc/video-getParameters.html b/LayoutTests/webrtc/video-getParameters.html
>index 3b269080b3dcf4d96f2510942a0e5e708adf1b94..e3a2931a296a68007f55f06e745a3c524856adb7 100644
>--- a/LayoutTests/webrtc/video-getParameters.html
>+++ b/LayoutTests/webrtc/video-getParameters.html
>@@ -51,7 +51,7 @@ promise_test((test) => {
>         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,"dtx":"disabled","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 parameters");
>     });
> }, "Basic video stats");
>         </script>

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Attachments on bug 189307: 348946