Replace voe_conference_test.

webrtc/audio/BUILD.gn

 # Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
 #
 # Use of this source code is governed by a BSD-style license
 # that can be found in the LICENSE file in the root of the source
 # tree. An additional intellectual property rights grant can be found
 # in the file PATENTS.  All contributing project authors may
 # be found in the AUTHORS file in the root of the source tree.
 
 import("../webrtc.gni")
 if (is_android) {
@@ skipping 40 matching lines @@
     "../modules/pacing:pacing",
     "../modules/remote_bitrate_estimator:remote_bitrate_estimator",
     "../modules/rtp_rtcp:rtp_rtcp",
     "../rtc_base:rtc_base_approved",
     "../rtc_base:rtc_task_queue",
     "../system_wrappers",
     "../voice_engine",
   ]
 }
 if (rtc_include_tests) {
+  rtc_source_set("audio_end_to_end_test") {
+    testonly = true
+
+    sources = [
+      "test/audio_end_to_end_test.cc",
+      "test/audio_end_to_end_test.h",
+    ]
+    deps = [
+      ":audio",
+      "../system_wrappers:system_wrappers",
+      "../test:fake_audio_device",
+      "../test:test_common",
+      "../test:test_support",
+    ]
+
+    if (!build_with_chromium && is_clang) {
+      # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
+      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
+    }
+  }
+
   rtc_source_set("audio_tests") {
     testonly = true
 
     # Skip restricting visibility on mobile platforms since the tests on those
     # gets additional generated targets which would require many lines here to
     # cover (which would be confusing to read and hard to maintain).
     if (!is_android && !is_ios) {
       visibility = [ "..:video_engine_tests" ]
     }
 
     # TODO(kjellander): Remove (bugs.webrtc.org/6828)
     # This needs remote_bitrate_estimator to be moved to webrtc/api first.
     check_includes = false
 
     sources = [
       "audio_receive_stream_unittest.cc",
       "audio_send_stream_unittest.cc",
       "audio_state_unittest.cc",
       "time_interval_unittest.cc",
     ]
     deps = [
       ":audio",
+      ":audio_end_to_end_test",
       "../api:mock_audio_mixer",
       "../call:rtp_receiver",
       "../modules/audio_device:mock_audio_device",
       "../modules/audio_mixer:audio_mixer_impl",
       "../modules/congestion_controller:congestion_controller",
       "../modules/congestion_controller:mock_congestion_controller",
       "../modules/pacing:pacing",
       "../rtc_base:rtc_base_approved",
       "../rtc_base:rtc_task_queue",
       "../test:test_common",
       "../test:test_support",
       "utility:utility_tests",
       "//testing/gmock",
       "//testing/gtest",
     ]
 
+    if (!rtc_use_memcheck) {
+      # This test is timing dependent, which rules out running on memcheck bots.
+      sources += [ "test/audio_stats_test.cc" ]
+    }
+
     if (!build_with_chromium && is_clang) {
       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
     }
   }
 
   if (rtc_enable_protobuf) {
     rtc_test("low_bandwidth_audio_test") {
       testonly = true
 
       sources = [
         "test/low_bandwidth_audio_test.cc",
-        "test/low_bandwidth_audio_test.h",
       ]
 
       deps = [
+        ":audio_end_to_end_test",
         "../common_audio",
         "../rtc_base:rtc_base_approved",
         "../system_wrappers",
         "../test:fake_audio_device",
         "../test:test_common",
         "../test:test_main",
         "//testing/gmock",
         "//testing/gtest",
       ]
       if (is_android) {
@@ skipping 42 matching lines @@
     data = [
       "//resources/voice_engine/audio_dtx16.wav",
     ]
 
     if (!build_with_chromium && is_clang) {
       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
     }
   }
 }