GN: New conventions, default target and refactorings

webrtc/modules/audio_coding/BUILD.gn

 # Copyright (c) 2014 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("../../build/webrtc.gni")
 import("audio_coding.gni")
@@ skipping 895 matching lines @@
     "neteq/tools/neteq_test.h",
   ]
 
   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_include_tests) {
+  group("audio_coding_tests") {
+    testonly = true
+    public_deps = [
+      ":RTPchange",
+      ":RTPencode",
+      ":RTPjitter",
+      ":RTPtimeshift",
+      ":acm_receive_test",
+      ":acm_send_test",
+      ":audio_classifier_test",
+      ":audio_codec_speed_tests",
+      ":audio_decoder_unittests",
+      ":audio_decoder_unittests",
+      ":delay_test",
+      ":g711_test",
+      ":g722_test",
+      ":ilbc_test",
+      ":insert_packet_with_timing",
+      ":isac_api_test",
+      ":isac_fix_test",
+      ":isac_switch_samprate_test",
+      ":isac_test",
+      ":neteq_ilbc_quality_test",
+      ":neteq_isac_quality_test",
+      ":neteq_opus_quality_test",
+      ":neteq_pcmu_quality_test",
+      ":neteq_speed_test",
+      ":rtp_analyze",
+      ":rtpcat",
+      ":webrtc_opus_fec_test",
+    ]
+    if (rtc_enable_protobuf) {
+      public_deps += [ ":neteq_rtpplay" ]
+    }
+  }
+
   rtc_source_set("acm_receive_test") {
     testonly = true
     sources = [
       "acm2/acm_receive_test.cc",
       "acm2/acm_receive_test.h",
     ]
 
     defines = audio_coding_defines
 
     deps = audio_coding_deps + [
@@ skipping 95 matching lines @@
 
     defines = neteq_defines
 
     deps += audio_coding_deps
     deps += [
       ":audio_decoder_interface",
       ":isac",
       ":isac_fix",
       ":neteq",
       ":neteq_unittest_tools",
-      "../../common_audio/",
+      "../../common_audio",
       "../../test:test_support_main",
       "//testing/gtest",
     ]
 
     data = audio_decoder_unittests_resources
 
     if (is_android) {
       deps += [ "//testing/android/native_test:native_test_native_code" ]
       shard_timeout = 900
     }
@@ skipping 79 matching lines @@
     if (is_android) {
       deps += [ "//testing/android/native_test:native_test_native_code" ]
       shard_timeout = 900
     }
 
     deps += [
       ":isac_fix",
       ":webrtc_opus",
       "../../system_wrappers:system_wrappers_default",
       "../../test:test_support_main",
-      "../audio_processing/",
+      "../audio_processing",
       "//testing/gtest",
     ]
   }
 
   rtc_source_set("neteq_test_support") {
     testonly = true
     sources = [
       "neteq/tools/neteq_external_decoder_test.cc",
       "neteq/tools/neteq_external_decoder_test.h",
       "neteq/tools/neteq_performance_test.cc",
@@ skipping 494 matching lines @@
       "../../test:test_support_main",
       "//testing/gtest",
     ]
 
     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" ]
     }
   }
 }