Prevent Opus DTX from generating intermittent noise during silence

webrtc/voice_engine/voice_engine.gyp

 # Copyright (c) 2011 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.
 
 {
   'includes': [
@@ skipping 10 matching lines @@
         '<(webrtc_root)/modules/modules.gyp:audio_conference_mixer',
         '<(webrtc_root)/modules/modules.gyp:audio_device',
         '<(webrtc_root)/modules/modules.gyp:audio_processing',
         '<(webrtc_root)/modules/modules.gyp:bitrate_controller',
         '<(webrtc_root)/modules/modules.gyp:media_file',
         '<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
         '<(webrtc_root)/modules/modules.gyp:webrtc_utility',
         '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
         '<(webrtc_root)/webrtc.gyp:rtc_event_log',
       ],
+      'export_dependent_settings': [
+        '<(webrtc_root)/modules/modules.gyp:audio_coding_module',
+      ],
       'sources': [
         'include/voe_audio_processing.h',
         'include/voe_base.h',
         'include/voe_codec.h',
         'include/voe_dtmf.h',
         'include/voe_errors.h',
         'include/voe_external_media.h',
         'include/voe_file.h',
         'include/voe_hardware.h',
         'include/voe_neteq_stats.h',
@@ skipping 106 matching lines @@
           'type': 'executable',
           'dependencies': [
             'voice_engine',
             '<(DEPTH)/testing/gmock.gyp:gmock',
             '<(DEPTH)/testing/gtest.gyp:gtest',
             '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
             '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
             '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
             '<(webrtc_root)/test/test.gyp:channel_transport',
             '<(webrtc_root)/test/test.gyp:test_support',
+            '<(webrtc_root)/test/webrtc_test_common.gyp:webrtc_test_common',
             '<(webrtc_root)/webrtc.gyp:rtc_event_log',
            ],
           'sources': [
             'test/auto_test/automated_mode.cc',
             'test/auto_test/extended/agc_config_test.cc',
             'test/auto_test/extended/ec_metrics_test.cc',
             'test/auto_test/fakes/conference_transport.cc',
             'test/auto_test/fakes/conference_transport.h',
             'test/auto_test/fakes/loudest_filter.cc',
             'test/auto_test/fakes/loudest_filter.h',
@@ skipping 20 matching lines @@
             'test/auto_test/standard/rtp_rtcp_before_streaming_test.cc',
             'test/auto_test/standard/rtp_rtcp_extensions.cc',
             'test/auto_test/standard/rtp_rtcp_test.cc',
             'test/auto_test/standard/voe_base_misc_test.cc',
             'test/auto_test/standard/video_sync_test.cc',
             'test/auto_test/standard/volume_test.cc',
             'test/auto_test/resource_manager.cc',
             'test/auto_test/voe_conference_test.cc',
             'test/auto_test/voe_cpu_test.cc',
             'test/auto_test/voe_cpu_test.h',
+            'test/auto_test/voe_output_test.cc',
             'test/auto_test/voe_standard_test.cc',
             'test/auto_test/voe_standard_test.h',
             'test/auto_test/voe_stress_test.cc',
             'test/auto_test/voe_stress_test.h',
             'test/auto_test/voe_test_defines.h',
             'test/auto_test/voe_test_interface.h',
           ],
           'conditions': [
             ['OS=="android"', {
               # some tests are not supported on android yet, exclude these tests.
@@ skipping 70 matching lines @@
               'sources': [
                 'voe_auto_test.isolate',
               ],
             },
           ],
         }],
       ],  # conditions
     }], # include_tests
   ], # conditions
 }