Import updated WebIDL for RTCRtpSendParameters and related types

Extracts RTCRtpSendParameters from RTCRtpParameters.
Extracts RTCRtpEncodingParameters from RTCRtpCodingParameters.
Adds required to many fields and update tests accordindly.

Bug: 803494
Change-Id: I416c0884abf15ab8d55ce5ae481f15ac8012c73d
Reviewed-on: https://chromium-review.googlesource.com/1114844Reviewed-by: Kentaro Hara <haraken@chromium.org>
Reviewed-by: Henrik Boström <hbos@chromium.org>
Commit-Queue: Florent Castelli <orphis@chromium.org>
Cr-Commit-Position: refs/heads/master@{#570820}

third_party/WebKit/LayoutTests/fast/peerconnection/RTCRtpSender-setParameters-expected.txt

@@ -4,65 +4,59 @@ FAIL setParameters() with large maxBitrate changes getParameters() returned valu
 PASS setParameters() with multiple calls after single getParameters()
 PASS setParameters() fails without getParameters()
 PASS video setParameters() check for transactionId modification
-PASS video setParameters() check for transactionId removed
 PASS video setParameters() check for codecs remove one
 PASS video setParameters() check for codecs add one
 PASS video setParameters() check for codecs reorder
-PASS video setParameters() check for codecs removed
 PASS video setParameters() check for codecs.payloadtype modification
-PASS video setParameters() check for codecs.payloadtype removed
 PASS video setParameters() check for codecs.mimeType modification
-PASS video setParameters() check for codecs.mimeType remove
 PASS video setParameters() check for codecs.clockRate modification
-PASS video setParameters() check for codecs.clockRate removed
 PASS video setParameters() check for codecs.channels modification
 PASS video setParameters() check for codecs.channels removed
 PASS video setParameters() check for codecs.sdpFmtpLine modification
 PASS video setParameters() check for codecs.sdpFmtpLine removed
-PASS video setParameters() check for encodings removed
 PASS video setParameters() check for encodings remove one
 PASS video setParameters() check for encodings add one
 FAIL video setParameters() check for encodings.rid modification assert_unreached: Should have rejected: undefined Reached unreachable code
 PASS video setParameters() check for headerExtensions remove one
 PASS video setParameters() check for headerExtensions add one
 PASS video setParameters() check for headerExtensions.uri modification
-PASS video setParameters() check for headerExtensions.uri removed
 PASS video setParameters() check for headerExtensions.id modification
-PASS video setParameters() check for headerExtensions.id removed
 PASS video setParameters() check for headerExtensions.encrypted modification
 PASS video setParameters() check for headerExtensions.encrypted removed
 PASS video setParameters() check for rtcp.cname modification
 PASS video setParameters() check for rtcp.reducedSize modification
 PASS audio setParameters() check for transactionId modification
-PASS audio setParameters() check for transactionId removed
 PASS audio setParameters() check for codecs remove one
 PASS audio setParameters() check for codecs add one
 PASS audio setParameters() check for codecs reorder
-PASS audio setParameters() check for codecs removed
 PASS audio setParameters() check for codecs.payloadtype modification
-PASS audio setParameters() check for codecs.payloadtype removed
 PASS audio setParameters() check for codecs.mimeType modification
-PASS audio setParameters() check for codecs.mimeType remove
 PASS audio setParameters() check for codecs.clockRate modification
-PASS audio setParameters() check for codecs.clockRate removed
 PASS audio setParameters() check for codecs.channels modification
 PASS audio setParameters() check for codecs.channels removed
 PASS audio setParameters() check for codecs.sdpFmtpLine modification
 PASS audio setParameters() check for codecs.sdpFmtpLine removed
-PASS audio setParameters() check for encodings removed
 PASS audio setParameters() check for encodings remove one
 PASS audio setParameters() check for encodings add one
 FAIL audio setParameters() check for encodings.rid modification assert_unreached: Should have rejected: undefined Reached unreachable code
 PASS audio setParameters() check for headerExtensions remove one
 PASS audio setParameters() check for headerExtensions add one
 PASS audio setParameters() check for headerExtensions.uri modification
-PASS audio setParameters() check for headerExtensions.uri removed
 PASS audio setParameters() check for headerExtensions.id modification
-PASS audio setParameters() check for headerExtensions.id removed
 PASS audio setParameters() check for headerExtensions.encrypted modification
 PASS audio setParameters() check for headerExtensions.encrypted removed
 PASS audio setParameters() check for rtcp.cname modification
 PASS audio setParameters() check for rtcp.reducedSize modification
+PASS video setParameters() check for transactionId is required
+PASS video setParameters() check for codecs is required
+PASS video setParameters() check for codecs.payloadType is required
+PASS video setParameters() check for codecs.mimeType is required
+PASS video setParameters() check for codecs.clockRate is required
+PASS video setParameters() check for encodings is required
+PASS video setParameters() check for headerExtensions is required
+PASS video setParameters() check for headerExtensions.uri is required
+PASS video setParameters() check for headerExtensions.id is required
+PASS video setParameters() check for rtcp is required
 PASS setParameters() set low maxBitrate value
 Harness: the test ran to completion.
 

third_party/WebKit/LayoutTests/fast/peerconnection/RTCRtpSender-setParameters.html

@@ -81,7 +81,13 @@ promise_test(async (t) => {
     new DOMException(
       "getParameters() needs to be called before setParameters().",
       "InvalidStateError"),
-    videoSender.setParameters({ rtcp: {}, headerExtensions: []}));
+    videoSender.setParameters({
+      transactionId: "",
+      codecs: [],
+      encodings: [],
+      rtcp: {},
+      headerExtensions: []
+    }));
 }, 'setParameters() fails without getParameters()');
 
 /**
@@ -97,10 +103,6 @@ var readOnlyParameters = [
     name: "transactionId modification",
     transform: (p) => { p.transactionId = "foo"; },
   },
-  {
-    name: "transactionId removed",
-    transform: (p) => { delete p.transactionId; },
-  },
   {
     name: "codecs remove one",
     transform: (p) => { p.codecs.pop(); },
@@ -119,36 +121,20 @@ var readOnlyParameters = [
   },
   {
     name: "codecs reorder",
-    transform: (p) => { p.codecs.push(p.codecs.splice(-1, 1)); },
-  },
-  {
-    name: "codecs removed",
-    transform: (p) => { delete p.codecs; },
+    transform: (p) => { p.codecs.concat(p.codecs.splice(0, 1)); },
   },
   {
     name: "codecs.payloadtype modification",
     transform: (p) => { p.codecs[0].payloadType = 1337; },
   },
-  {
-    name: "codecs.payloadtype removed",
-    transform: (p) => { delete p.codecs[0].payloadType; },
-  },
   {
     name: "codecs.mimeType modification",
     transform: (p) => { p.codecs[0].mimeType = "mime"; },
   },
-  {
-    name: "codecs.mimeType remove",
-    transform: (p) => { delete p.codecs[0].mimeType; },
-  },
   {
     name: "codecs.clockRate modification",
     transform: (p) => { p.codecs[0].clockRate = 42; },
   },
-  {
-    name: "codecs.clockRate removed",
-    transform: (p) => { delete p.codecs[0].clockRate; },
-  },
   {
     name: "codecs.channels modification",
     transform: (p) => { p.codecs[0].channels = 16; },
@@ -167,10 +153,6 @@ var readOnlyParameters = [
     transform: (p) => { delete p.codecs[0].sdpFmtpLine; },
     check: (p) => { return p.codecs[0].sdpFmtpLine !== undefined; },
   },
-  {
-    name: "encodings removed",
-    transform: (p) => { delete p.encodings; },
-  },
   {
     name: "encodings remove one",
     transform: (p) => { p.encodings.pop(); },
@@ -203,11 +185,6 @@ var readOnlyParameters = [
     transform: (p) => { p.headerExtensions[0].uri = "uri"; },
     check: (p) => { return p.headerExtensions.length != 0; },
   },
-  {
-    name: "headerExtensions.uri removed",
-    transform: (p) => { delete p.headerExtensions[0].uri; },
-    check: (p) => { return p.headerExtensions.length != 0; },
-  },
   {
     name: "headerExtensions.id modification",
     transform: (p) => {
@@ -217,11 +194,6 @@ var readOnlyParameters = [
       return p.headerExtensions.length != 0;
     },
   },
-  {
-    name: "headerExtensions.id removed",
-    transform: (p) => { delete p.headerExtensions[0].id; },
-    check: (p) => { return p.headerExtensions.length != 0; },
-  },
   {
     name: "headerExtensions.encrypted modification",
     transform: (p) => {
@@ -293,6 +265,67 @@ for (testCase of readOnlyParameters) {
   'audio setParameters() check for ' + testCase.name);
 }
 
+var requiredParameters = [
+  {
+    name: "transactionId is required",
+    transform: (p) => { delete p.transactionId; },
+  },
+  {
+    name: "codecs is required",
+    transform: (p) => { delete p.codecs; },
+  },
+  {
+    name: "codecs.payloadType is required",
+    transform: (p) => { delete p.codecs[0].payloadType; },
+  },
+  {
+    name: "codecs.mimeType is required",
+    transform: (p) => { delete p.codecs[0].mimeType; },
+  },
+  {
+    name: "codecs.clockRate is required",
+    transform: (p) => { delete p.codecs[0].clockRate; },
+  },
+  {
+    name: "encodings is required",
+    transform: (p) => { delete p.encodings; },
+  },
+  {
+    name: "headerExtensions is required",
+    transform: (p) => { delete p.headerExtensions; },
+  },
+  {
+    name: "headerExtensions.uri is required",
+    transform: (p) => { delete p.headerExtensions[0].uri; },
+  },
+  {
+    name: "headerExtensions.id is required",
+    transform: (p) => { delete p.headerExtensions[0].id; },
+  },
+  {
+    name: "rtcp is required",
+    transform: (p) => { delete p.rtcp; },
+  },
+];
+
+for (testCase of requiredParameters) {
+  promise_test((async (name, transform, check, t) => {
+    let pc1 = new RTCPeerConnection();
+    let pc2 = new RTCPeerConnection();
+
+    let stream = await navigator.mediaDevices.getUserMedia({ video: true }, 1);
+    let videoSender = pc1.addTrack(stream.getVideoTracks()[0], stream);
+    await doSignalingHandshake(pc1, pc2);
+
+    let videoParameters = videoSender.getParameters();
+    transform(videoParameters);
+      return promise_rejects(t,
+        new TypeError(),
+        videoSender.setParameters(videoParameters));
+  }).bind(undefined, testCase.name, testCase.transform, testCase.check),
+  'video setParameters() check for ' + testCase.name);
+}
+
 promise_test(async function() {
   let pc1 = new RTCPeerConnection();
   let pc2 = new RTCPeerConnection();

third_party/blink/renderer/modules/modules_idl_files.gni

@@ -581,9 +581,11 @@ modules_dictionary_idl_files =
           "peerconnection/rtc_peer_connection_ice_event_init.idl",
           "peerconnection/rtc_rtcp_parameters.idl",
           "peerconnection/rtc_rtp_codec_parameters.idl",
+          "peerconnection/rtc_rtp_coding_parameters.idl",
           "peerconnection/rtc_rtp_encoding_parameters.idl",
           "peerconnection/rtc_rtp_header_extension_parameters.idl",
           "peerconnection/rtc_rtp_parameters.idl",
+          "peerconnection/rtc_rtp_send_parameters.idl",
           "peerconnection/rtc_session_description_init.idl",
           "peerconnection/rtc_track_event_init.idl",
           "permissions/clipboard_permission_descriptor.idl",

third_party/blink/renderer/modules/peerconnection/rtc_rtp_codec_parameters.idl

@@ -4,9 +4,9 @@
 
 // https://w3c.github.io/webrtc-pc/#rtcrtpcodecparameters*
 dictionary RTCRtpCodecParameters {
-    octet          payloadType;
-    DOMString      mimeType;
-    unsigned long  clockRate;
-    unsigned short channels;
-    DOMString      sdpFmtpLine;
+    required octet          payloadType;
+    required DOMString      mimeType;
+    required unsigned long  clockRate;
+             unsigned short channels;
+             DOMString      sdpFmtpLine;
 };
\ No newline at end of file

third_party/blink/renderer/modules/peerconnection/rtc_rtp_coding_parameters.idl

+// Copyright 2018 The Chromium Authors. All rights reserved.
+// Use of this source code is governed by a BSD-style license that can be
+// found in the LICENSE file.
+
+// https://w3c.github.io/webrtc-pc/#rtcrtpcodingparameters
+dictionary RTCRtpCodingParameters {
+    //TODO(orphis): Missing rid. https://crbug.com/857034
+};
\ No newline at end of file

third_party/blink/renderer/modules/peerconnection/rtc_rtp_encoding_parameters.idl

@@ -11,8 +11,13 @@ enum RTCPriorityType {
 };
 
 // https://w3c.github.io/webrtc-pc/#rtcrtpencodingparameters*
-dictionary RTCRtpEncodingParameters {
+dictionary RTCRtpEncodingParameters : RTCRtpCodingParameters {
+    //TODO(orphis): Missing codecPayloadType. https://crbug.com/857042
+    //TODO(orphis): Missing dtx. https://crbug.com/857044
     boolean         active = true;
     RTCPriorityType priority = "low";
+    //TODO(orphis): Missing ptime. https://crbug.com/857046
     unsigned long   maxBitrate;
+    //TODO(orphis): Missing maxFramerate. https://crbug.com/857047
+    //TODO(orphis): Missing scaleResolutionDownBy. https://crbug.com/857048
 };
\ No newline at end of file

third_party/blink/renderer/modules/peerconnection/rtc_rtp_header_extension_parameters.idl

@@ -4,7 +4,7 @@
 
 // https://w3c.github.io/webrtc-pc/#rtcrtpheaderextensionparameters*
 dictionary RTCRtpHeaderExtensionParameters {
-    DOMString      uri;
-    unsigned short id;
-    boolean        encrypted;
+    required DOMString      uri;
+    required unsigned short id;
+             boolean        encrypted = false;
 };
\ No newline at end of file

third_party/blink/renderer/modules/peerconnection/rtc_rtp_parameters.idl

@@ -4,9 +4,7 @@
 
 // https://w3c.github.io/webrtc-pc/#rtcrtpparameters*
 dictionary RTCRtpParameters {
-    DOMString                                 transactionId;
-    sequence<RTCRtpEncodingParameters>        encodings;
     required sequence<RTCRtpHeaderExtensionParameters> headerExtensions;
-    required RTCRtcpParameters                rtcp;
-    sequence<RTCRtpCodecParameters>           codecs;
+    required RTCRtcpParameters                         rtcp;
+    required sequence<RTCRtpCodecParameters>           codecs;
 };
\ No newline at end of file

third_party/blink/renderer/modules/peerconnection/rtc_rtp_send_parameters.idl

+// Copyright 2018 The Chromium Authors. All rights reserved.
+// Use of this source code is governed by a BSD-style license that can be
+// found in the LICENSE file.
+
+// https://w3c.github.io/webrtc-pc/#rtcsendrtpparameters
+dictionary RTCRtpSendParameters : RTCRtpParameters {
+    required DOMString                          transactionId;
+    required sequence<RTCRtpEncodingParameters> encodings;
+    //TODO(orphis): Missing degradationPrerence. https://crbug.com/857041
+};
\ No newline at end of file

third_party/blink/renderer/modules/peerconnection/rtc_rtp_sender.cc

@@ -82,8 +82,8 @@ class SetParametersRequest : public RTCVoidRequestScriptPromiseResolverImpl {
   Member<RTCRtpSender> sender_;
 };
 
-bool HasInvalidModification(const RTCRtpParameters& parameters,
-                            const RTCRtpParameters& new_parameters) {
+bool HasInvalidModification(const RTCRtpSendParameters& parameters,
+                            const RTCRtpSendParameters& new_parameters) {
   if (parameters.hasTransactionId() != new_parameters.hasTransactionId() ||
       (parameters.hasTransactionId() &&
        parameters.transactionId() != new_parameters.transactionId())) {
@@ -215,7 +215,7 @@ double PriorityToDouble(const WTF::String& priority) {
 
 std::tuple<std::vector<webrtc::RtpEncodingParameters>,
            webrtc::DegradationPreference>
-ToRtpParameters(const RTCRtpParameters& parameters) {
+ToRtpParameters(const RTCRtpSendParameters& parameters) {
   std::vector<webrtc::RtpEncodingParameters> encodings;
   if (parameters.hasEncodings()) {
     encodings.reserve(parameters.encodings().size());
@@ -275,9 +275,9 @@ ScriptPromise RTCRtpSender::replaceTrack(ScriptState* script_state,
   return promise;
 }
 
-void RTCRtpSender::getParameters(RTCRtpParameters& parameters) {
+void RTCRtpSender::getParameters(RTCRtpSendParameters& parameters) {
   // TODO(orphis): Forward missing fields from the WebRTC library:
-  // rtcp, headerExtensions, degradationPreference
+  // degradationPreference
   std::unique_ptr<webrtc::RtpParameters> webrtc_parameters =
       sender_->GetParameters();
 
@@ -340,8 +340,9 @@ void RTCRtpSender::getParameters(RTCRtpParameters& parameters) {
   last_returned_parameters_ = parameters;
 }
 
-ScriptPromise RTCRtpSender::setParameters(ScriptState* script_state,
-                                          const RTCRtpParameters& parameters) {
+ScriptPromise RTCRtpSender::setParameters(
+    ScriptState* script_state,
+    const RTCRtpSendParameters& parameters) {
   ScriptPromiseResolver* resolver = ScriptPromiseResolver::Create(script_state);
   ScriptPromise promise = resolver->Promise();
 

third_party/blink/renderer/modules/peerconnection/rtc_rtp_sender.h

@@ -8,7 +8,7 @@
 #include "third_party/blink/public/platform/web_rtc_rtp_sender.h"
 #include "third_party/blink/renderer/bindings/core/v8/script_promise.h"
 #include "third_party/blink/renderer/modules/mediastream/media_stream.h"
-#include "third_party/blink/renderer/modules/peerconnection/rtc_rtp_parameters.h"
+#include "third_party/blink/renderer/modules/peerconnection/rtc_rtp_send_parameters.h"
 #include "third_party/blink/renderer/platform/bindings/script_wrappable.h"
 #include "third_party/blink/renderer/platform/heap/garbage_collected.h"
 #include "third_party/blink/renderer/platform/heap/member.h"
@@ -20,7 +20,6 @@ namespace blink {
 class MediaStreamTrack;
 class RTCDTMFSender;
 class RTCPeerConnection;
-class RTCRtpParameters;
 
 // https://w3c.github.io/webrtc-pc/#rtcrtpsender-interface
 class RTCRtpSender final : public ScriptWrappable {
@@ -37,8 +36,8 @@ class RTCRtpSender final : public ScriptWrappable {
   MediaStreamTrack* track();
   ScriptPromise replaceTrack(ScriptState*, MediaStreamTrack*);
   RTCDTMFSender* dtmf();
-  void getParameters(RTCRtpParameters&);
-  ScriptPromise setParameters(ScriptState*, const RTCRtpParameters&);
+  void getParameters(RTCRtpSendParameters&);
+  ScriptPromise setParameters(ScriptState*, const RTCRtpSendParameters&);
   ScriptPromise getStats(ScriptState*);
 
   WebRTCRtpSender* web_sender();
@@ -59,7 +58,7 @@ class RTCRtpSender final : public ScriptWrappable {
   String kind_;
   Member<RTCDTMFSender> dtmf_;
   MediaStreamVector streams_;
-  base::Optional<RTCRtpParameters> last_returned_parameters_;
+  base::Optional<RTCRtpSendParameters> last_returned_parameters_;
 };
 
 }  // namespace blink

third_party/blink/renderer/modules/peerconnection/rtc_rtp_sender.idl

@@ -6,8 +6,8 @@
 [Exposed=Window]
 interface RTCRtpSender {
     readonly attribute MediaStreamTrack? track;
-    [RuntimeEnabled=RTCRtpSenderParameters, CallWith=ScriptState] Promise<void> setParameters(optional RTCRtpParameters parameters);
-    [RuntimeEnabled=RTCRtpSenderParameters] RTCRtpParameters getParameters();
+    [RuntimeEnabled=RTCRtpSenderParameters, CallWith=ScriptState] Promise<void> setParameters(optional RTCRtpSendParameters parameters);
+    [RuntimeEnabled=RTCRtpSenderParameters] RTCRtpSendParameters getParameters();
     [Measure, CallWith=ScriptState] Promise<void> replaceTrack(MediaStreamTrack? withTrack);
     [Measure] readonly attribute RTCDTMFSender? dtmf;
     [CallWith=ScriptState] Promise<RTCStatsReport> getStats();