From 529f83c521c6726017bed9da16664063b0041eaf Mon Sep 17 00:00:00 2001
From: ehmaldonado <ehmaldonado@webrtc.org>
Date: Wed, 27 Jul 2016 08:14:32 -0700
Subject: [PATCH] Add webrtc_perf_tests to BUILD.gn

Updated the sources in audio_processing:audioproc_test_utils to match the configuration on
"webrtc/modules/audio_processing/audio_processing_tests.gypi"

Removed audio_buffer_tools from modules_unittests to match the gyp file.

BUG=webrtc:6041

Review-Url: https://codereview.webrtc.org/2178963002
Cr-Commit-Position: refs/heads/master@{#13541}
---
 webrtc/BUILD.gn                          | 66 ++++++++++++++++++++++++
 webrtc/modules/BUILD.gn                  |  2 -
 webrtc/modules/audio_processing/BUILD.gn |  2 +
 3 files changed, 68 insertions(+), 2 deletions(-)

diff --git a/webrtc/BUILD.gn b/webrtc/BUILD.gn
index 3e197ed096..be0e88e442 100644
--- a/webrtc/BUILD.gn
+++ b/webrtc/BUILD.gn
@@ -619,6 +619,72 @@ if (rtc_include_tests) {
     }
   }
 
+  source_set("video_quality_test") {
+    testonly = true
+    configs += [ ":common_config" ]
+    public_configs = [ ":common_inherited_config" ]
+    sources = [
+      "video/video_quality_test.cc",
+      "video/video_quality_test.h",
+    ]
+    deps = [
+      ":webrtc",
+      "system_wrappers",
+      "//testing/gtest",
+    ]
+    if (!is_android) {
+      deps += [ "modules/video_capture:video_capture_internal_impl" ]
+    }
+    if (is_clang) {
+      # Suppress warnings from the Chromium Clang plugin.
+      # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
+      configs -= [ "//build/config/clang:find_bad_constructs" ]
+    }
+  }
+
+  test("webrtc_perf_tests") {
+    testonly = true
+    configs += [
+      ":common_config",
+      ":rtc_unittests_config",
+    ]
+    public_configs = [ ":common_inherited_config" ]
+    sources = [
+      "call/call_perf_tests.cc",
+      "call/rampup_tests.cc",
+      "call/rampup_tests.h",
+      "modules/audio_coding/neteq/test/neteq_performance_unittest.cc",
+      "modules/audio_processing/audio_processing_performance_unittest.cc",
+      "modules/audio_processing/level_controller/level_controller_complexity_unittest.cc",
+      "modules/remote_bitrate_estimator/remote_bitrate_estimators_test.cc",
+      "video/full_stack.cc",
+    ]
+    deps = [
+      ":video_quality_test",
+      ":webrtc",
+      "modules/audio_coding:neteq_test_support",
+      "modules/audio_processing",
+      "modules/audio_processing:audioproc_test_utils",
+      "modules/remote_bitrate_estimator:bwe_simulator",
+      "modules/rtp_rtcp",
+      "test:channel_transport",
+      "test:test_common",
+      "test:test_main",
+      "test:test_renderer",
+      "voice_engine",
+      "//testing/gmock",
+      "//testing/gtest",
+    ]
+    if (is_android) {
+      deps += [ "//testing/android/native_test:native_test_native_code" ]
+    }
+    if (is_clang) {
+      # Suppress warnings from the Chromium Clang plugin.
+      # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
+      configs -= [ "//build/config/clang:find_bad_constructs" ]
+    }
+  }
+
   test("webrtc_nonparallel_tests") {
     testonly = true
     configs += [
diff --git a/webrtc/modules/BUILD.gn b/webrtc/modules/BUILD.gn
index 1b1d9630aa..08c4bdb329 100644
--- a/webrtc/modules/BUILD.gn
+++ b/webrtc/modules/BUILD.gn
@@ -322,8 +322,6 @@ if (rtc_include_tests) {
         "audio_processing/high_pass_filter_unittest.cc",
         "audio_processing/level_estimator_unittest.cc",
         "audio_processing/noise_suppression_unittest.cc",
-        "audio_processing/test/audio_buffer_tools.cc",
-        "audio_processing/test/audio_buffer_tools.h",
         "audio_processing/test/bitexactness_tools.cc",
         "audio_processing/test/bitexactness_tools.h",
         "audio_processing/test/debug_dump_replayer.cc",
diff --git a/webrtc/modules/audio_processing/BUILD.gn b/webrtc/modules/audio_processing/BUILD.gn
index 9c1674a102..7ef0a77f3b 100644
--- a/webrtc/modules/audio_processing/BUILD.gn
+++ b/webrtc/modules/audio_processing/BUILD.gn
@@ -319,6 +319,8 @@ if (rtc_include_tests) {
   source_set("audioproc_test_utils") {
     testonly = true
     sources = [
+      "test/audio_buffer_tools.cc",
+      "test/audio_buffer_tools.h",
       "test/test_utils.cc",
       "test/test_utils.h",
     ]