Index: webrtc/modules/rtp_rtcp/BUILD.gn |
diff --git a/webrtc/modules/rtp_rtcp/BUILD.gn b/webrtc/modules/rtp_rtcp/BUILD.gn |
index e806f070a1afc47a4b75b26c85bee833a82e27b9..9b236218b3f986da9ca49d7ce3142958c2c7570f 100644 |
--- a/webrtc/modules/rtp_rtcp/BUILD.gn |
+++ b/webrtc/modules/rtp_rtcp/BUILD.gn |
@@ -210,4 +210,93 @@ if (rtc_include_tests) { |
"//webrtc/test:test_main", |
] |
} # test_packet_masks_metrics |
+ |
+ rtc_source_set("rtp_rtcp_unittests") { |
+ testonly = true |
+ sources = [ |
+ "mocks/mock_rtp_rtcp.h", |
+ "source/byte_io_unittest.cc", |
+ "source/fec_test_helper.cc", |
+ "source/fec_test_helper.h", |
+ "source/flexfec_header_reader_writer_unittest.cc", |
+ "source/flexfec_receiver_unittest.cc", |
+ "source/flexfec_sender_unittest.cc", |
+ "source/nack_rtx_unittest.cc", |
+ "source/packet_loss_stats_unittest.cc", |
+ "source/playout_delay_oracle_unittest.cc", |
+ "source/receive_statistics_unittest.cc", |
+ "source/remote_ntp_time_estimator_unittest.cc", |
+ "source/rtcp_packet/app_unittest.cc", |
+ "source/rtcp_packet/bye_unittest.cc", |
+ "source/rtcp_packet/common_header_unittest.cc", |
+ "source/rtcp_packet/compound_packet_unittest.cc", |
+ "source/rtcp_packet/dlrr_unittest.cc", |
+ "source/rtcp_packet/extended_jitter_report_unittest.cc", |
+ "source/rtcp_packet/extended_reports_unittest.cc", |
+ "source/rtcp_packet/fir_unittest.cc", |
+ "source/rtcp_packet/nack_unittest.cc", |
+ "source/rtcp_packet/pli_unittest.cc", |
+ "source/rtcp_packet/rapid_resync_request_unittest.cc", |
+ "source/rtcp_packet/receiver_report_unittest.cc", |
+ "source/rtcp_packet/remb_unittest.cc", |
+ "source/rtcp_packet/report_block_unittest.cc", |
+ "source/rtcp_packet/rpsi_unittest.cc", |
+ "source/rtcp_packet/rrtr_unittest.cc", |
+ "source/rtcp_packet/sdes_unittest.cc", |
+ "source/rtcp_packet/sender_report_unittest.cc", |
+ "source/rtcp_packet/sli_unittest.cc", |
+ "source/rtcp_packet/target_bitrate_unittest.cc", |
+ "source/rtcp_packet/tmmbn_unittest.cc", |
+ "source/rtcp_packet/tmmbr_unittest.cc", |
+ "source/rtcp_packet/transport_feedback_unittest.cc", |
+ "source/rtcp_packet/voip_metric_unittest.cc", |
+ "source/rtcp_packet_unittest.cc", |
+ "source/rtcp_receiver_unittest.cc", |
+ "source/rtcp_sender_unittest.cc", |
+ "source/rtcp_utility_unittest.cc", |
+ "source/rtp_fec_unittest.cc", |
+ "source/rtp_format_h264_unittest.cc", |
+ "source/rtp_format_vp8_test_helper.cc", |
+ "source/rtp_format_vp8_test_helper.h", |
+ "source/rtp_format_vp8_unittest.cc", |
+ "source/rtp_format_vp9_unittest.cc", |
+ "source/rtp_header_extension_unittest.cc", |
+ "source/rtp_packet_history_unittest.cc", |
+ "source/rtp_packet_unittest.cc", |
+ "source/rtp_payload_registry_unittest.cc", |
+ "source/rtp_rtcp_impl_unittest.cc", |
+ "source/rtp_sender_unittest.cc", |
+ "source/rtp_utility_unittest.cc", |
+ "source/time_util_unittest.cc", |
+ "source/ulpfec_generator_unittest.cc", |
+ "source/ulpfec_header_reader_writer_unittest.cc", |
+ "source/ulpfec_receiver_unittest.cc", |
+ "source/vp8_partition_aggregator_unittest.cc", |
+ "test/testAPI/test_api.cc", |
+ "test/testAPI/test_api.h", |
+ "test/testAPI/test_api_audio.cc", |
+ "test/testAPI/test_api_rtcp.cc", |
+ "test/testAPI/test_api_video.cc", |
+ ] |
+ deps = [ |
+ ":rtp_rtcp", |
+ "../..:webrtc_common", |
+ "../../api:transport_api", |
+ "../../base:rtc_base_approved", |
+ "../../common_video:common_video", |
+ "../../system_wrappers:system_wrappers", |
+ "../../test:field_trial", |
+ "../../test:rtp_test_utils", |
+ "../../test:test_common", |
+ "../../test:test_support", |
+ "//testing/gmock", |
+ ] |
+ |
+ # TODO(jschuh): bugs.webrtc.org/1348: fix this warning. |
+ configs += [ "//build/config/compiler:no_size_t_to_int_warning" ] |
+ 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" ] |
+ } |
+ } |
} |