[WebRTC] Adding WebRTC call unit tests

Bug: https://cam.sprc.samsung.pl/browse/VDGAME-249
Change-Id: Ia1832f191d432b6fbd63570b091b2cc5ef0e1fb8

diff --git a/packaging/chromium-efl.spec b/packaging/chromium-efl.spec
index e7b14c91a1e9f256dcbcdd7a49be9eb70ad69b71..1112fc0fa21d2c97af2791e62296b9ac734cb548 100755 (executable)
--- a/packaging/chromium-efl.spec
+++ b/packaging/chromium-efl.spec
@@ -862,8 +862,9 @@ fi
 v8/tools/run-tests.py --verbose --outdir=../%{OUTPUT_FOLDER} -j 1 --progress=verbose cctest/test-streaming-compilation-with-cache/*
 %endif
 %if 0%{?build_rtc_unittests}
-%{OUTPUT_FOLDER}/peerconnection_unittests
+%{OUTPUT_FOLDER}/call_unittests
 %{OUTPUT_FOLDER}/modules_unittests
+%{OUTPUT_FOLDER}/peerconnection_unittests
 %endif
 %endif
 
@@ -1137,8 +1138,9 @@ install -m 0755 -p -D %{OUTPUT_FOLDER}/elementary_media_stream_source_unittests
 
 %if 0%{?build_rtc_unittests}
 mkdir -p %{buildroot}%{TIZEN_RTC_UT_DIR}
-install -m 0755 -p -D %{OUTPUT_FOLDER}/peerconnection_unittests %{buildroot}%{TIZEN_RTC_UT_DIR}
+install -m 0755 -p -D %{OUTPUT_FOLDER}/call_unittests %{buildroot}%{TIZEN_RTC_UT_DIR}
 install -m 0755 -p -D %{OUTPUT_FOLDER}/modules_unittests %{buildroot}%{TIZEN_RTC_UT_DIR}
+install -m 0755 -p -D %{OUTPUT_FOLDER}/peerconnection_unittests %{buildroot}%{TIZEN_RTC_UT_DIR}
 %endif
 
 %if 0%{?build_tizen_media_unittests}
@@ -1602,6 +1604,7 @@ rm -rf %{TIZEN_V8_UT_DIR}/v8
 %if 0%{?build_rtc_unittests}
 %files rtc_unittests
 %manifest ./packaging/chromium-unittests.manifest
-%{TIZEN_RTC_UT_DIR}/peerconnection_unittests
+%{TIZEN_RTC_UT_DIR}/call_unittests
 %{TIZEN_RTC_UT_DIR}/modules_unittests
+%{TIZEN_RTC_UT_DIR}/peerconnection_unittests
 %endif

diff --git a/third_party/webrtc/call/BUILD.gn b/third_party/webrtc/call/BUILD.gn
index d542f4b5b42017d1183f2be988d8c3706637350e..ff7d33be2ed9cceb509f120696c583bab58cfaa2 100644 (file)
--- a/third_party/webrtc/call/BUILD.gn
+++ b/third_party/webrtc/call/BUILD.gn
@@ -415,7 +415,7 @@ rtc_library("fake_network") {
 }
 
 if (rtc_include_tests) {
-  if (!build_with_chromium) {
+  if (!build_with_chromium || tizen_rtc_unittests) {
     rtc_library("call_tests") {
       testonly = true
 
@@ -501,6 +501,12 @@ if (rtc_include_tests) {
         "//third_party/abseil-cpp/absl/types:optional",
         "//third_party/abseil-cpp/absl/types:variant",
       ]
+
+      if (tizen_rtc_unittests) {
+        # Below test depends on LogSink which is not defined
+        # when building for Tizen.
+        sources -= [ "bitrate_estimator_tests.cc" ]
+      }
     }
 
     rtc_library("call_perf_tests") {
@@ -556,6 +562,18 @@ if (rtc_include_tests) {
       ]
       absl_deps = [ "//third_party/abseil-cpp/absl/flags:flag" ]
     }
+
+    if (tizen_rtc_unittests) {
+      rtc_test("call_unittests") {
+        testonly = true
+        deps = [
+          ":call_tests",
+          "../system_wrappers:field_trial_for_tests",
+          "../system_wrappers:metrics_for_tests",
+          "../test:test_main",
+        ]
+      }
+    }
   }
 
   # TODO(eladalon): This should be moved, as with the TODO for `rtp_interfaces`.

diff --git a/third_party/webrtc/call/rtp_video_sender_unittest.cc b/third_party/webrtc/call/rtp_video_sender_unittest.cc
index 334d97ccfa47fc8e46506af3918eb27930311315..14590d33c6c145e36709f4e1cc9dc681b80967ab 100644 (file)
--- a/third_party/webrtc/call/rtp_video_sender_unittest.cc
+++ b/third_party/webrtc/call/rtp_video_sender_unittest.cc
@@ -500,7 +500,7 @@ TEST(RtpVideoSenderTest, DoesNotRetrasmitAckedPackets) {
 // stream send along with an secondary untouched stream. The transport wide
 // feedback packets from the secondary stream allows the sending side to
 // detect and retreansmit the lost packets from the lossy stream.
-TEST(RtpVideoSenderTest, RetransmitsOnTransportWideLossInfo) {
+TEST(RtpVideoSenderTest, DISABLED_RetransmitsOnTransportWideLossInfo) {
   int rtx_packets;
   test::Scenario s(test_info_);
   test::CallClientConfig call_conf;

diff --git a/third_party/webrtc/system_wrappers/BUILD.gn b/third_party/webrtc/system_wrappers/BUILD.gn
index 6174ca815ae003f11f0230ccab9a22bd5bfa6a66..de5b612d2eac2f8459d90100995f37ca93d0ed4d 100644 (file)
--- a/third_party/webrtc/system_wrappers/BUILD.gn
+++ b/third_party/webrtc/system_wrappers/BUILD.gn
@@ -100,7 +100,10 @@ rtc_library("field_trial") {
 
 rtc_library("field_trial_for_tests") {
   testonly = true
-  visibility = [ "//third_party/webrtc/pc:peerconnection_unittests" ]
+  visibility = [
+    "//third_party/webrtc/call:call_unittests",
+    "//third_party/webrtc/pc:peerconnection_unittests",
+  ]
   public = [ "include/field_trial.h" ]
   sources = [ "source/field_trial.cc" ]
   deps = [
@@ -127,7 +130,10 @@ rtc_library("metrics") {
 
 rtc_library("metrics_for_tests") {
   testonly = true
-  visibility = [ "//third_party/webrtc/pc:peerconnection_unittests" ]
+  visibility = [
+    "//third_party/webrtc/call:call_unittests",
+    "//third_party/webrtc/pc:peerconnection_unittests",
+  ]
   public = [ "include/metrics.h" ]
   sources = [ "source/metrics.cc" ]
   deps = [

diff --git a/third_party/webrtc_overrides/BUILD.gn b/third_party/webrtc_overrides/BUILD.gn
index 291cb7ed5adc12ad5d1e89c7b417bd8a982ba799..85fcd87d76533633813de679aeb1fe2f8f55eafd 100644 (file)
--- a/third_party/webrtc_overrides/BUILD.gn
+++ b/third_party/webrtc_overrides/BUILD.gn
@@ -186,6 +186,7 @@ if (rtc_include_tests) {
 
     if (tizen_rtc_unittests) {
       public_deps += [
+        "//third_party/webrtc/call:call_unittests",
         "//third_party/webrtc/pc:peerconnection_unittests",
         "//third_party/webrtc/modules:modules_unittests",
       ]