diff --git a/logging/BUILD.gn b/logging/BUILD.gn index ee2f455980..9d1acf019c 100644 --- a/logging/BUILD.gn +++ b/logging/BUILD.gn @@ -14,11 +14,11 @@ if (is_android) { } group("logging") { - public_deps = [ + deps = [ ":rtc_event_log_impl", ] if (rtc_enable_protobuf) { - public_deps += [ ":rtc_event_log_parser" ] + deps += [ ":rtc_event_log_parser" ] } } @@ -139,13 +139,17 @@ if (rtc_enable_protobuf) { "rtc_event_log/rtc_event_log_parser.h", ] - public_deps = [ + deps = [ ":rtc_event_log_api", ":rtc_event_log_proto", "..:webrtc_common", + "../call:video_stream_api", "../modules/audio_coding:audio_network_adaptor", "../modules/remote_bitrate_estimator:remote_bitrate_estimator", "../modules/rtp_rtcp", + "../modules/rtp_rtcp:rtp_rtcp_format", + "../rtc_base:protobuf_utils", + "../rtc_base:rtc_base_approved", "../system_wrappers", ] @@ -153,12 +157,6 @@ if (rtc_enable_protobuf) { # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] } - deps = [ - "../call:video_stream_api", - "../modules/rtp_rtcp:rtp_rtcp_format", - "../rtc_base:protobuf_utils", - "../rtc_base:rtc_base_approved", - ] } if (rtc_include_tests) { @@ -177,8 +175,10 @@ if (rtc_enable_protobuf) { "rtc_event_log/rtc_event_log_unittest_helper.h", ] deps = [ + ":rtc_event_log_api", ":rtc_event_log_impl", ":rtc_event_log_parser", + ":rtc_event_log_proto", "../api:libjingle_peerconnection_api", "../call", "../call:call_interfaces", @@ -206,7 +206,9 @@ if (rtc_enable_protobuf) { ":rtc_event_log_api", ":rtc_event_log_impl", ":rtc_event_log_parser", + "../modules/rtp_rtcp", "../modules/rtp_rtcp:rtp_rtcp_format", + "../rtc_base:protobuf_utils", "../rtc_base:rtc_base_approved", "../system_wrappers:field_trial_default", "../system_wrappers:metrics_default", @@ -228,8 +230,10 @@ if (rtc_enable_protobuf) { ":rtc_event_log_api", ":rtc_event_log_impl", ":rtc_event_log_parser", + "../:webrtc_common", "../call:video_stream_api", "../modules/rtp_rtcp:rtp_rtcp_format", + "../rtc_base:protobuf_utils", "../rtc_base:rtc_base_approved", # TODO(kwiberg): Remove this dependency. diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn index 0754f74d3f..557247ffd3 100644 --- a/modules/audio_coding/BUILD.gn +++ b/modules/audio_coding/BUILD.gn @@ -1194,6 +1194,7 @@ if (rtc_enable_protobuf) { ":neteq_tools_minimal", "../../logging:rtc_event_log_parser", "../../rtc_base:rtc_base_approved", + "../rtp_rtcp", "../rtp_rtcp:rtp_rtcp_format", ] public_deps = [ diff --git a/rtc_tools/BUILD.gn b/rtc_tools/BUILD.gn index 49e4568c86..1fbc627e09 100644 --- a/rtc_tools/BUILD.gn +++ b/rtc_tools/BUILD.gn @@ -223,12 +223,15 @@ if (!build_with_chromium) { defines = [ "ENABLE_RTC_EVENT_LOG" ] deps = [ ":chart_proto", + "../:webrtc_common", "../call:call_interfaces", "../call:video_stream_api", + "../logging:rtc_event_log_api", "../logging:rtc_event_log_impl", "../logging:rtc_event_log_parser", "../modules:module_api", "../modules/audio_coding:ana_debug_dump_proto", + "../modules/audio_coding:audio_network_adaptor", "../modules/audio_coding:neteq_tools", "../modules/rtp_rtcp:rtp_rtcp_format", "../rtc_base:rtc_base_approved", @@ -263,6 +266,7 @@ if (rtc_include_tests) { deps = [ ":event_log_visualizer_utils", "../logging:rtc_event_log_parser", + "../rtc_base:protobuf_utils", "../rtc_base:rtc_base_approved", "../test:field_trial", "../test:test_support",