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LeOSium_webview/LeOS/patches/Add-webRTC-site-settings.patch

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From: uazo <uazo@users.noreply.github.com>
Date: Fri, 6 May 2022 14:27:17 +0000
Subject: Add webRTC site settings
Requires patch: Content-settings-infrastructure.patch
Original License: GPL-2.0-or-later - https://spdx.org/licenses/GPL-2.0-or-later.html
License: GPL-3.0-only - https://spdx.org/licenses/GPL-3.0-only.html
---
.../impl/BromiteWebRTCContentSetting.java | 93 +++++++++++++++++++
.../bromite_content_settings/webrtc.grdp | 27 ++++++
.../bromite_content_settings/webrtc.inc | 22 +++++
.../bromite_content_settings/WEBRTC.inc | 1 +
.../peer_connection_dependency_factory.cc | 6 ++
.../peerconnection/rtc_rtp_receiver.cc | 10 ++
.../modules/peerconnection/rtc_rtp_sender.cc | 10 ++
7 files changed, 169 insertions(+)
create mode 100644 components/browser_ui/site_settings/android/java/src/org/chromium/components/browser_ui/site_settings/impl/BromiteWebRTCContentSetting.java
create mode 100644 components/browser_ui/strings/bromite_content_settings/webrtc.grdp
create mode 100644 components/content_settings/core/browser/bromite_content_settings/webrtc.inc
create mode 100644 components/content_settings/core/common/bromite_content_settings/WEBRTC.inc
diff --git a/components/browser_ui/site_settings/android/java/src/org/chromium/components/browser_ui/site_settings/impl/BromiteWebRTCContentSetting.java b/components/browser_ui/site_settings/android/java/src/org/chromium/components/browser_ui/site_settings/impl/BromiteWebRTCContentSetting.java
new file mode 100644
--- /dev/null
+++ b/components/browser_ui/site_settings/android/java/src/org/chromium/components/browser_ui/site_settings/impl/BromiteWebRTCContentSetting.java
@@ -0,0 +1,93 @@
+/*
+ This file is part of Bromite.
+
+ Bromite is free software: you can redistribute it and/or modify
+ it under the terms of the GNU General Public License as published by
+ the Free Software Foundation, either version 3 of the License, or
+ (at your option) any later version.
+
+ Bromite is distributed in the hope that it will be useful,
+ but WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ GNU General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with Bromite. If not, see <https://www.gnu.org/licenses/>.
+*/
+
+package org.chromium.components.browser_ui.site_settings.impl;
+
+import org.chromium.components.browser_ui.site_settings.R;
+
+import org.chromium.components.browser_ui.site_settings.BromiteCustomContentSetting;
+import org.chromium.components.browser_ui.site_settings.ContentSettingsResources;
+import org.chromium.components.browser_ui.site_settings.SiteSettingsCategory;
+import org.chromium.components.content_settings.ContentSettingValues;
+import org.chromium.components.content_settings.ContentSettingsType;
+import org.chromium.content_public.browser.BrowserContextHandle;
+
+import androidx.annotation.Nullable;
+import androidx.preference.Preference;
+import androidx.preference.PreferenceScreen;
+
+import java.util.ArrayList;
+
+public class BromiteWebRTCContentSetting extends BromiteCustomContentSetting {
+ public BromiteWebRTCContentSetting() {
+ super(/*contentSettingsType*/ ContentSettingsType.WEBRTC,
+ /*defaultEnabledValue*/ ContentSettingValues.ALLOW,
+ /*defaultDisabledValue*/ ContentSettingValues.BLOCK,
+ /*allowException*/ true,
+ /*preferenceKey*/ "webrtc",
+ /*profilePrefKey*/ "webrtc");
+ }
+
+ @Override
+ public ContentSettingsResources.ResourceItem getResourceItem() {
+ return new ContentSettingsResources.ResourceItem(
+ /*icon*/ R.drawable.web_asset,
+ /*title*/ R.string.webrtc_permission_title,
+ /*defaultEnabledValue*/ getDefaultEnabledValue(),
+ /*defaultDisabledValue*/ getDefaultDisabledValue(),
+ /*enabledSummary*/ R.string.website_settings_category_webrtc_enabled,
+ /*disabledSummary*/ R.string.website_settings_category_webrtc_disabled);
+ }
+
+ @Override
+ public int getCategorySummary(@Nullable @ContentSettingValues int value) {
+ switch (value) {
+ case ContentSettingValues.ALLOW:
+ return R.string.website_settings_category_webrtc_enabled;
+ case ContentSettingValues.BLOCK:
+ return R.string.website_settings_category_webrtc_disabled;
+ default:
+ // this will cause a runtime exception
+ return 0;
+ }
+ }
+
+ @Override
+ public int getCategoryDescription() {
+ return R.string.settings_site_settings_webrtc_description;
+ }
+
+ @Override
+ public boolean requiresTriStateContentSetting() {
+ return false;
+ }
+
+ @Override
+ public boolean showOnlyDescriptions() {
+ return true;
+ }
+
+ @Override
+ public int getAddExceptionDialogMessage() {
+ return R.string.website_settings_category_webrtc_enabled;
+ }
+
+ @Override
+ public @Nullable Boolean considerException(SiteSettingsCategory category, @ContentSettingValues int value) {
+ return value != ContentSettingValues.BLOCK;
+ }
+}
diff --git a/components/browser_ui/strings/bromite_content_settings/webrtc.grdp b/components/browser_ui/strings/bromite_content_settings/webrtc.grdp
new file mode 100644
--- /dev/null
+++ b/components/browser_ui/strings/bromite_content_settings/webrtc.grdp
@@ -0,0 +1,27 @@
+<?xml version="1.0" encoding="utf-8"?>
+<grit-part>
+ <message name="IDS_SITE_SETTINGS_TYPE_WEBRTC" desc="The label used for webrtc site settings controls." formatter_data="android_java">
+ WebRTC
+ </message>
+ <message name="IDS_SITE_SETTINGS_TYPE_WEBRTC_MID_SENTENCE" desc="The label used for webrtc site settings controls when used mid-sentence." formatter_data="android_java">
+ webRTC
+ </message>
+ <message name="IDS_SETTINGS_SITE_SETTINGS_WEBRTC_DESCRIPTION" desc="Description of the WebRTC content setting." formatter_data="android_java">
+ Enable WebRTC, a JavaScript API for real-time communication and direct peer-to-peer communication
+ </message>
+ <message name="IDS_WEBRTC_PERMISSION_TITLE" desc="Title of the permission to use webrtc [CHAR-LIMIT=32]" formatter_data="android_java">
+ WebRTC
+ </message>
+ <message name="IDS_WEBSITE_SETTINGS_CATEGORY_WEBRTC_ENABLED" desc="Summary text explaining that webrtc is full enabled." formatter_data="android_java">
+ Enabled
+ </message>
+ <message name="IDS_WEBSITE_SETTINGS_CATEGORY_WEBRTC_DISABLED" desc="Summary text explaining that webrtc is full disabled." formatter_data="android_java">
+ Disabled
+ </message>
+ <message name="IDS_SETTINGS_SITE_SETTINGS_WEBRTC_ALLOWED_EXCEPTIONS" desc="Label for the allowed exceptions site list of the WebRTC setting.">
+ Allowed to use WebRTC
+ </message>
+ <message name="IDS_SETTINGS_SITE_SETTINGS_WEBRTC_BLOCKED_EXCEPTIONS" desc="Label for the blocked exceptions site list of the WebRTC setting.">
+ Not allowed to use WebRTC
+ </message>
+</grit-part>
diff --git a/components/content_settings/core/browser/bromite_content_settings/webrtc.inc b/components/content_settings/core/browser/bromite_content_settings/webrtc.inc
new file mode 100644
--- /dev/null
+++ b/components/content_settings/core/browser/bromite_content_settings/webrtc.inc
@@ -0,0 +1,22 @@
+ Register(ContentSettingsType::WEBRTC, "webrtc", CONTENT_SETTING_BLOCK,
+ WebsiteSettingsInfo::SYNCABLE,
+ /*allowlisted_schemes=*/{},
+ /*valid_settings=*/{CONTENT_SETTING_ALLOW,
+ CONTENT_SETTING_BLOCK},
+ WebsiteSettingsInfo::TOP_ORIGIN_ONLY_SCOPE,
+ WebsiteSettingsRegistry::ALL_PLATFORMS,
+ ContentSettingsInfo::INHERIT_IN_INCOGNITO,
+ ContentSettingsInfo::EXCEPTIONS_ON_SECURE_AND_INSECURE_ORIGINS);
+
+ content_settings::WebsiteSettingsRegistry::GetInstance()
+ ->GetMutable(ContentSettingsType::WEBRTC)
+ ->set_show_into_info_page()
+ .set_desktop_ui()
+ .set_is_renderer_content_setting()
+ .set_title_ui(IDS_SITE_SETTINGS_TYPE_WEBRTC)
+ .set_description_ui(IDS_SETTINGS_SITE_SETTINGS_WEBRTC_DESCRIPTION)
+ .set_allowed_ui(IDS_WEBSITE_SETTINGS_CATEGORY_WEBRTC_ENABLED)
+ .set_blocked_ui(IDS_WEBSITE_SETTINGS_CATEGORY_WEBRTC_DISABLED)
+ .set_allowed_exceptions_ui(IDS_SETTINGS_SITE_SETTINGS_WEBRTC_ALLOWED_EXCEPTIONS)
+ .set_blocked_exceptions_ui(IDS_SETTINGS_SITE_SETTINGS_WEBRTC_BLOCKED_EXCEPTIONS)
+ .set_mid_sentence_ui(IDS_SITE_SETTINGS_TYPE_WEBRTC_MID_SENTENCE);
diff --git a/components/content_settings/core/common/bromite_content_settings/WEBRTC.inc b/components/content_settings/core/common/bromite_content_settings/WEBRTC.inc
new file mode 100644
--- /dev/null
+++ b/components/content_settings/core/common/bromite_content_settings/WEBRTC.inc
@@ -0,0 +1 @@
+ WEBRTC,
diff --git a/third_party/blink/renderer/modules/peerconnection/peer_connection_dependency_factory.cc b/third_party/blink/renderer/modules/peerconnection/peer_connection_dependency_factory.cc
--- a/third_party/blink/renderer/modules/peerconnection/peer_connection_dependency_factory.cc
+++ b/third_party/blink/renderer/modules/peerconnection/peer_connection_dependency_factory.cc
@@ -37,6 +37,7 @@
#include "third_party/blink/public/common/peerconnection/webrtc_ip_handling_policy.h"
#include "third_party/blink/public/platform/modules/webrtc/webrtc_logging.h"
#include "third_party/blink/public/platform/platform.h"
+#include "third_party/blink/public/platform/web_content_settings_client.h"
#include "third_party/blink/public/platform/web_url.h"
#include "third_party/blink/public/web/modules/mediastream/media_stream_video_source.h"
#include "third_party/blink/public/web/web_document.h"
@@ -781,6 +782,11 @@ PeerConnectionDependencyFactory::CreatePortAllocator(
// origin.
WebRTCIPHandlingPolicy policy =
GetWebRTCIPHandlingPolicy(webrtc_ip_handling_policy);
+ blink::WebContentSettingsClient* settings = web_frame->GetContentSettingsClient();
+ if (settings && settings->AllowContentSetting(ContentSettingsType::WEBRTC, false)) {
+ policy = kDefault;
+ }
+
switch (policy) {
// TODO(guoweis): specify the flag of disabling local candidate
// collection when webrtc is updated.
diff --git a/third_party/blink/renderer/modules/peerconnection/rtc_rtp_receiver.cc b/third_party/blink/renderer/modules/peerconnection/rtc_rtp_receiver.cc
--- a/third_party/blink/renderer/modules/peerconnection/rtc_rtp_receiver.cc
+++ b/third_party/blink/renderer/modules/peerconnection/rtc_rtp_receiver.cc
@@ -12,6 +12,8 @@
#include "third_party/blink/public/common/privacy_budget/identifiable_surface.h"
#include "third_party/blink/public/common/privacy_budget/identifiable_token_builder.h"
#include "third_party/blink/public/platform/modules/webrtc/webrtc_logging.h"
+#include "third_party/blink/public/platform/web_content_settings_client.h"
+#include "third_party/blink/public/web/web_local_frame.h"
#include "third_party/blink/renderer/bindings/core/v8/script_promise_resolver.h"
#include "third_party/blink/renderer/bindings/modules/v8/v8_rtc_insertable_streams.h"
#include "third_party/blink/renderer/bindings/modules/v8/v8_rtc_rtcp_parameters.h"
@@ -255,6 +257,14 @@ RTCRtpCapabilities* RTCRtpReceiver::getCapabilities(ScriptState* state,
if (kind != "audio" && kind != "video")
return nullptr;
+ LocalDOMWindow* window = To<LocalDOMWindow>(ExecutionContext::From(state));
+ auto* web_frame =
+ static_cast<WebLocalFrame*>(WebFrame::FromCoreFrame(window->GetFrame()));
+ blink::WebContentSettingsClient* settings = web_frame->GetContentSettingsClient();
+ if (settings && !settings->AllowContentSetting(ContentSettingsType::WEBRTC, false)) {
+ return nullptr;
+ }
+
RTCRtpCapabilities* capabilities = RTCRtpCapabilities::Create();
capabilities->setCodecs(HeapVector<Member<RTCRtpCodecCapability>>());
capabilities->setHeaderExtensions(
diff --git a/third_party/blink/renderer/modules/peerconnection/rtc_rtp_sender.cc b/third_party/blink/renderer/modules/peerconnection/rtc_rtp_sender.cc
--- a/third_party/blink/renderer/modules/peerconnection/rtc_rtp_sender.cc
+++ b/third_party/blink/renderer/modules/peerconnection/rtc_rtp_sender.cc
@@ -17,6 +17,8 @@
#include "third_party/blink/public/common/privacy_budget/identifiable_surface.h"
#include "third_party/blink/public/common/privacy_budget/identifiable_token_builder.h"
#include "third_party/blink/public/platform/modules/webrtc/webrtc_logging.h"
+#include "third_party/blink/public/platform/web_content_settings_client.h"
+#include "third_party/blink/public/web/web_local_frame.h"
#include "third_party/blink/renderer/bindings/core/v8/script_promise_resolver.h"
#include "third_party/blink/renderer/bindings/modules/v8/v8_rtc_insertable_streams.h"
#include "third_party/blink/renderer/bindings/modules/v8/v8_rtc_rtcp_parameters.h"
@@ -1004,6 +1006,14 @@ RTCRtpCapabilities* RTCRtpSender::getCapabilities(ScriptState* state,
if (kind != "audio" && kind != "video")
return nullptr;
+ LocalDOMWindow* window = To<LocalDOMWindow>(ExecutionContext::From(state));
+ auto* web_frame =
+ static_cast<WebLocalFrame*>(WebFrame::FromCoreFrame(window->GetFrame()));
+ blink::WebContentSettingsClient* settings = web_frame->GetContentSettingsClient();
+ if (settings && !settings->AllowContentSetting(ContentSettingsType::WEBRTC, false)) {
+ return nullptr;
+ }
+
RTCRtpCapabilities* capabilities = RTCRtpCapabilities::Create();
capabilities->setCodecs(HeapVector<Member<RTCRtpCodecCapability>>());
capabilities->setHeaderExtensions(
--
2.25.1
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