Relanding "Setting up an RTP input fuzzer for NetEq"

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 850 matching lines @@
       deps += [ ":isac_fix" ]
     } else {
       defines += [ "WEBRTC_CODEC_ISAC" ]
       deps += [ ":isac" ]
     }
     defines += [ "WEBRTC_CODEC_G722" ]
     deps += [ ":g722" ]
   }
 }
 
+# Although providing only test support, this target must be outside of the
+# rtc_include_tests conditional. The reason is that it supports fuzzer tests
+# that ultimately are built and run as a part of the Chromium ecosystem, which
+# does not set the rtc_include_tests flag.
+rtc_source_set("neteq_test_minimal") {
+  testonly = true
+  sources = [
+    "neteq/tools/encode_neteq_input.cc",
+    "neteq/tools/encode_neteq_input.h",
+    "neteq/tools/neteq_test.cc",
+    "neteq/tools/neteq_test.h",
+  ]
+
+  if (is_clang) {
+    # Suppress warnings from the Chromium Clang plugins (bugs.webrtc.org/163).
+    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
+  }
+}
+
 if (rtc_include_tests) {
   rtc_source_set("acm_receive_test") {
     testonly = true
     sources = [
       "acm2/acm_receive_test_oldapi.cc",
       "acm2/acm_receive_test_oldapi.h",
     ]
 
     defines = audio_coding_defines
 
@@ skipping 268 matching lines @@
       "neteq/tools/constant_pcm_packet_source.h",
       "neteq/tools/fake_decode_from_file.cc",
       "neteq/tools/fake_decode_from_file.h",
       "neteq/tools/input_audio_file.cc",
       "neteq/tools/input_audio_file.h",
       "neteq/tools/neteq_input.h",
       "neteq/tools/neteq_packet_source_input.cc",
       "neteq/tools/neteq_packet_source_input.h",
       "neteq/tools/neteq_replacement_input.cc",
       "neteq/tools/neteq_replacement_input.h",
-      "neteq/tools/neteq_test.cc",
-      "neteq/tools/neteq_test.h",
       "neteq/tools/output_audio_file.h",
       "neteq/tools/output_wav_file.h",
       "neteq/tools/packet.cc",
       "neteq/tools/packet.h",
       "neteq/tools/packet_source.cc",
       "neteq/tools/packet_source.h",
       "neteq/tools/resample_input_audio_file.cc",
       "neteq/tools/resample_input_audio_file.h",
       "neteq/tools/rtp_file_source.cc",
       "neteq/tools/rtp_file_source.h",
       "neteq/tools/rtp_generator.cc",
       "neteq/tools/rtp_generator.h",
     ]
 
     public_configs = [ ":neteq_unittest_tools_config" ]
 
     if (is_clang) {
       # Suppress warnings from the Chromium Clang plugins (bugs.webrtc.org/163).
       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
     }
 
     deps = [
+      ":neteq_test_minimal",
       "../../common_audio",
       "../../test:rtp_test_utils",
       "../rtp_rtcp",
     ]
 
     if (rtc_enable_protobuf) {
       deps += [ ":rtc_event_log_source" ]
     }
   }
 
@@ skipping 404 matching lines @@
       "//testing/gtest",
     ]
 
     if (is_clang) {
       # Suppress warnings from Chrome's Clang plugins.
       # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
     }
   }
 }