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-rw-r--r--chromium/third_party/webrtc/examples/BUILD.gn18
-rw-r--r--chromium/third_party/webrtc/examples/androidapp/src/org/appspot/apprtc/AppRTCAudioManager.java4
-rw-r--r--chromium/third_party/webrtc/examples/androidnativeapi/BUILD.gn1
-rw-r--r--chromium/third_party/webrtc/examples/peerconnection/client/peer_connection_client.cc12
4 files changed, 25 insertions, 10 deletions
diff --git a/chromium/third_party/webrtc/examples/BUILD.gn b/chromium/third_party/webrtc/examples/BUILD.gn
index 4d6d14d0d99..ab3d5edfc9d 100644
--- a/chromium/third_party/webrtc/examples/BUILD.gn
+++ b/chromium/third_party/webrtc/examples/BUILD.gn
@@ -101,7 +101,7 @@ if (is_android) {
rtc_android_library("AppRTCMobile_javalib") {
testonly = true
- android_manifest_for_lint = "androidapp/AndroidManifest.xml"
+ android_manifest = "androidapp/AndroidManifest.xml"
sources = [
"androidapp/src/org/appspot/apprtc/AppRTCAudioManager.java",
@@ -180,10 +180,10 @@ if (is_android) {
"androidapp/res/layout/fragment_call.xml",
"androidapp/res/layout/fragment_hud.xml",
"androidapp/res/menu/connect_menu.xml",
- "androidapp/res/values/arrays.xml",
- "androidapp/res/values/strings.xml",
"androidapp/res/values-v17/styles.xml",
"androidapp/res/values-v21/styles.xml",
+ "androidapp/res/values/arrays.xml",
+ "androidapp/res/values/strings.xml",
"androidapp/res/xml/preferences.xml",
]
custom_package = "org.appspot.apprtc"
@@ -207,7 +207,10 @@ if (is_android) {
deps = [
":AppRTCMobile_javalib",
+ "../sdk/android:camera_java",
"../sdk/android:libjingle_peerconnection_java",
+ "../sdk/android:peerconnection_java",
+ "../sdk/android:video_api_java",
"../sdk/android:video_java",
"//third_party/android_support_test_runner:runner_java",
"//third_party/junit",
@@ -379,14 +382,18 @@ if (is_ios || (is_mac && target_cpu != "x86")) {
if (rtc_apprtcmobile_broadcast_extension) {
bundle_data("AppRTCMobileBroadcastUpload_extension_bundle") {
testonly = true
- public_deps = [ ":AppRTCMobileBroadcastUpload" ] # no-presubmit-check TODO(webrtc:8603)
+ public_deps = [ # no-presubmit-check TODO(webrtc:8603)
+ ":AppRTCMobileBroadcastUpload", # prevent code format
+ ]
sources = [ "$root_out_dir/AppRTCMobileBroadcastUpload.appex" ]
outputs = [ "{{bundle_contents_dir}}/Plugins/{{source_file_part}}" ]
}
bundle_data("AppRTCMobileBroadcastSetupUI_extension_bundle") {
testonly = true
- public_deps = [ ":AppRTCMobileBroadcastSetupUI" ] # no-presubmit-check TODO(webrtc:8603)
+ public_deps = [ # no-presubmit-check TODO(webrtc:8603)
+ ":AppRTCMobileBroadcastSetupUI", # prevent code format
+ ]
sources = [ "$root_out_dir/AppRTCMobileBroadcastSetupUI.appex" ]
outputs = [ "{{bundle_contents_dir}}/Plugins/{{source_file_part}}" ]
}
@@ -859,6 +866,7 @@ if (is_android) {
deps = [
":AppRTCMobile_javalib",
+ "../sdk/android:peerconnection_java",
"//base:base_java_test_support",
"//third_party/google-truth:google_truth_java",
]
diff --git a/chromium/third_party/webrtc/examples/androidapp/src/org/appspot/apprtc/AppRTCAudioManager.java b/chromium/third_party/webrtc/examples/androidapp/src/org/appspot/apprtc/AppRTCAudioManager.java
index 7ae3d838dd9..c32ab964ad1 100644
--- a/chromium/third_party/webrtc/examples/androidapp/src/org/appspot/apprtc/AppRTCAudioManager.java
+++ b/chromium/third_party/webrtc/examples/androidapp/src/org/appspot/apprtc/AppRTCAudioManager.java
@@ -185,8 +185,8 @@ public class AppRTCAudioManager {
// Note that, the sensor will not be active until start() has been called.
proximitySensor = AppRTCProximitySensor.create(context,
// This method will be called each time a state change is detected.
- // Example: user holds his hand over the device (closer than ~5 cm),
- // or removes his hand from the device.
+ // Example: user holds their hand over the device (closer than ~5 cm),
+ // or removes their hand from the device.
this ::onProximitySensorChangedState);
Log.d(TAG, "defaultAudioDevice: " + defaultAudioDevice);
diff --git a/chromium/third_party/webrtc/examples/androidnativeapi/BUILD.gn b/chromium/third_party/webrtc/examples/androidnativeapi/BUILD.gn
index 9c114e859c4..2b2c71b9e8a 100644
--- a/chromium/third_party/webrtc/examples/androidnativeapi/BUILD.gn
+++ b/chromium/third_party/webrtc/examples/androidnativeapi/BUILD.gn
@@ -16,6 +16,7 @@ if (is_android) {
deps = [
":resources",
"//modules/audio_device:audio_device_java",
+ "//rtc_base:base_java",
"//sdk/android:camera_java",
"//sdk/android:surfaceviewrenderer_java",
"//sdk/android:video_api_java",
diff --git a/chromium/third_party/webrtc/examples/peerconnection/client/peer_connection_client.cc b/chromium/third_party/webrtc/examples/peerconnection/client/peer_connection_client.cc
index 3203941672e..a463ceed46c 100644
--- a/chromium/third_party/webrtc/examples/peerconnection/client/peer_connection_client.cc
+++ b/chromium/third_party/webrtc/examples/peerconnection/client/peer_connection_client.cc
@@ -43,9 +43,15 @@ rtc::AsyncSocket* CreateClientSocket(int family) {
} // namespace
PeerConnectionClient::PeerConnectionClient()
- : callback_(NULL), resolver_(NULL), state_(NOT_CONNECTED), my_id_(-1) {}
-
-PeerConnectionClient::~PeerConnectionClient() {}
+ : MessageHandler(false),
+ callback_(NULL),
+ resolver_(NULL),
+ state_(NOT_CONNECTED),
+ my_id_(-1) {}
+
+PeerConnectionClient::~PeerConnectionClient() {
+ rtc::Thread::Current()->Clear(this);
+}
void PeerConnectionClient::InitSocketSignals() {
RTC_DCHECK(control_socket_.get() != NULL);