| # 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("../../webrtc.gni") | 
 |  | 
 | rtc_source_set("rtp_rtcp_format") { | 
 |   public = [ | 
 |     "include/rtp_cvo.h", | 
 |     "include/rtp_header_extension_map.h", | 
 |     "include/rtp_rtcp_defines.h", | 
 |     "source/byte_io.h", | 
 |     "source/rtcp_packet.h", | 
 |     "source/rtcp_packet/app.h", | 
 |     "source/rtcp_packet/bye.h", | 
 |     "source/rtcp_packet/common_header.h", | 
 |     "source/rtcp_packet/compound_packet.h", | 
 |     "source/rtcp_packet/dlrr.h", | 
 |     "source/rtcp_packet/extended_jitter_report.h", | 
 |     "source/rtcp_packet/extended_reports.h", | 
 |     "source/rtcp_packet/fir.h", | 
 |     "source/rtcp_packet/nack.h", | 
 |     "source/rtcp_packet/pli.h", | 
 |     "source/rtcp_packet/psfb.h", | 
 |     "source/rtcp_packet/rapid_resync_request.h", | 
 |     "source/rtcp_packet/receiver_report.h", | 
 |     "source/rtcp_packet/remb.h", | 
 |     "source/rtcp_packet/report_block.h", | 
 |     "source/rtcp_packet/rrtr.h", | 
 |     "source/rtcp_packet/rtpfb.h", | 
 |     "source/rtcp_packet/sdes.h", | 
 |     "source/rtcp_packet/sender_report.h", | 
 |     "source/rtcp_packet/target_bitrate.h", | 
 |     "source/rtcp_packet/tmmb_item.h", | 
 |     "source/rtcp_packet/tmmbn.h", | 
 |     "source/rtcp_packet/tmmbr.h", | 
 |     "source/rtcp_packet/transport_feedback.h", | 
 |     "source/rtcp_packet/voip_metric.h", | 
 |     "source/rtp_header_extensions.h", | 
 |     "source/rtp_packet.h", | 
 |     "source/rtp_packet_received.h", | 
 |     "source/rtp_packet_to_send.h", | 
 |   ] | 
 |   sources = [ | 
 |     "include/rtp_rtcp_defines.cc", | 
 |     "source/rtcp_packet.cc", | 
 |     "source/rtcp_packet/app.cc", | 
 |     "source/rtcp_packet/bye.cc", | 
 |     "source/rtcp_packet/common_header.cc", | 
 |     "source/rtcp_packet/compound_packet.cc", | 
 |     "source/rtcp_packet/dlrr.cc", | 
 |     "source/rtcp_packet/extended_jitter_report.cc", | 
 |     "source/rtcp_packet/extended_reports.cc", | 
 |     "source/rtcp_packet/fir.cc", | 
 |     "source/rtcp_packet/nack.cc", | 
 |     "source/rtcp_packet/pli.cc", | 
 |     "source/rtcp_packet/psfb.cc", | 
 |     "source/rtcp_packet/rapid_resync_request.cc", | 
 |     "source/rtcp_packet/receiver_report.cc", | 
 |     "source/rtcp_packet/remb.cc", | 
 |     "source/rtcp_packet/report_block.cc", | 
 |     "source/rtcp_packet/rrtr.cc", | 
 |     "source/rtcp_packet/rtpfb.cc", | 
 |     "source/rtcp_packet/sdes.cc", | 
 |     "source/rtcp_packet/sender_report.cc", | 
 |     "source/rtcp_packet/target_bitrate.cc", | 
 |     "source/rtcp_packet/tmmb_item.cc", | 
 |     "source/rtcp_packet/tmmbn.cc", | 
 |     "source/rtcp_packet/tmmbr.cc", | 
 |     "source/rtcp_packet/transport_feedback.cc", | 
 |     "source/rtcp_packet/voip_metric.cc", | 
 |     "source/rtp_header_extension_map.cc", | 
 |     "source/rtp_header_extensions.cc", | 
 |     "source/rtp_packet.cc", | 
 |     "source/rtp_packet_received.cc", | 
 |   ] | 
 |  | 
 |   deps = [ | 
 |     "..:module_api", | 
 |     "../..:webrtc_common", | 
 |     "../../:typedefs", | 
 |     "../../api:array_view", | 
 |     "../../api:libjingle_peerconnection_api", | 
 |     "../../api:optional", | 
 |     "../../api:video_frame_api", | 
 |     "../../api/audio_codecs:audio_codecs_api", | 
 |     "../../common_video", | 
 |     "../../rtc_base:checks", | 
 |     "../../rtc_base:deprecation", | 
 |     "../../rtc_base:rtc_base_approved", | 
 |     "../../system_wrappers", | 
 |   ] | 
 | } | 
 |  | 
 | rtc_static_library("rtp_rtcp") { | 
 |   visibility = [ "*" ] | 
 |   sources = [ | 
 |     "include/flexfec_receiver.h", | 
 |     "include/flexfec_sender.h", | 
 |     "include/receive_statistics.h", | 
 |     "include/remote_ntp_time_estimator.h", | 
 |     "include/rtp_header_parser.h", | 
 |     "include/rtp_payload_registry.h", | 
 |     "include/rtp_receiver.h", | 
 |     "include/rtp_rtcp.h", | 
 |     "include/ulpfec_receiver.h", | 
 |     "source/dtmf_queue.cc", | 
 |     "source/dtmf_queue.h", | 
 |     "source/fec_private_tables_bursty.h", | 
 |     "source/fec_private_tables_random.h", | 
 |     "source/flexfec_header_reader_writer.cc", | 
 |     "source/flexfec_header_reader_writer.h", | 
 |     "source/flexfec_receiver.cc", | 
 |     "source/flexfec_sender.cc", | 
 |     "source/forward_error_correction.cc", | 
 |     "source/forward_error_correction.h", | 
 |     "source/forward_error_correction_internal.cc", | 
 |     "source/forward_error_correction_internal.h", | 
 |     "source/packet_loss_stats.cc", | 
 |     "source/packet_loss_stats.h", | 
 |     "source/playout_delay_oracle.cc", | 
 |     "source/playout_delay_oracle.h", | 
 |     "source/receive_statistics_impl.cc", | 
 |     "source/receive_statistics_impl.h", | 
 |     "source/remote_ntp_time_estimator.cc", | 
 |     "source/rtcp_nack_stats.cc", | 
 |     "source/rtcp_nack_stats.h", | 
 |     "source/rtcp_receiver.cc", | 
 |     "source/rtcp_receiver.h", | 
 |     "source/rtcp_sender.cc", | 
 |     "source/rtcp_sender.h", | 
 |     "source/rtp_format.cc", | 
 |     "source/rtp_format.h", | 
 |     "source/rtp_format_h264.cc", | 
 |     "source/rtp_format_h264.h", | 
 |     "source/rtp_format_video_generic.cc", | 
 |     "source/rtp_format_video_generic.h", | 
 |     "source/rtp_format_vp8.cc", | 
 |     "source/rtp_format_vp8.h", | 
 |     "source/rtp_format_vp9.cc", | 
 |     "source/rtp_format_vp9.h", | 
 |     "source/rtp_header_parser.cc", | 
 |     "source/rtp_packet_history.cc", | 
 |     "source/rtp_packet_history.h", | 
 |     "source/rtp_payload_registry.cc", | 
 |     "source/rtp_receiver_audio.cc", | 
 |     "source/rtp_receiver_audio.h", | 
 |     "source/rtp_receiver_impl.cc", | 
 |     "source/rtp_receiver_impl.h", | 
 |     "source/rtp_receiver_strategy.cc", | 
 |     "source/rtp_receiver_strategy.h", | 
 |     "source/rtp_receiver_video.cc", | 
 |     "source/rtp_receiver_video.h", | 
 |     "source/rtp_rtcp_config.h", | 
 |     "source/rtp_rtcp_impl.cc", | 
 |     "source/rtp_rtcp_impl.h", | 
 |     "source/rtp_sender.cc", | 
 |     "source/rtp_sender.h", | 
 |     "source/rtp_sender_audio.cc", | 
 |     "source/rtp_sender_audio.h", | 
 |     "source/rtp_sender_video.cc", | 
 |     "source/rtp_sender_video.h", | 
 |     "source/rtp_utility.cc", | 
 |     "source/rtp_utility.h", | 
 |     "source/time_util.cc", | 
 |     "source/time_util.h", | 
 |     "source/tmmbr_help.cc", | 
 |     "source/tmmbr_help.h", | 
 |     "source/ulpfec_generator.cc", | 
 |     "source/ulpfec_generator.h", | 
 |     "source/ulpfec_header_reader_writer.cc", | 
 |     "source/ulpfec_header_reader_writer.h", | 
 |     "source/ulpfec_receiver_impl.cc", | 
 |     "source/ulpfec_receiver_impl.h", | 
 |     "source/video_codec_information.h", | 
 |   ] | 
 |  | 
 |   if (rtc_enable_bwe_test_logging) { | 
 |     defines = [ "BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=1" ] | 
 |   } else { | 
 |     defines = [ "BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=0" ] | 
 |   } | 
 |  | 
 |   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" ] | 
 |   } | 
 |  | 
 |   deps = [ | 
 |     ":rtp_rtcp_format", | 
 |     "..:module_api", | 
 |     "../..:webrtc_common", | 
 |     "../../:typedefs", | 
 |     "../../api:array_view", | 
 |     "../../api:libjingle_peerconnection_api", | 
 |     "../../api:optional", | 
 |     "../../api:transport_api", | 
 |     "../../api/audio_codecs:audio_codecs_api", | 
 |     "../../common_video", | 
 |     "../../logging:rtc_event_audio", | 
 |     "../../logging:rtc_event_log_api", | 
 |     "../../logging:rtc_event_rtp_rtcp", | 
 |     "../../rtc_base:checks", | 
 |     "../../rtc_base:deprecation", | 
 |     "../../rtc_base:gtest_prod", | 
 |     "../../rtc_base:rate_limiter", | 
 |     "../../rtc_base:rtc_base_approved", | 
 |     "../../rtc_base:rtc_numerics", | 
 |     "../../rtc_base:sequenced_task_checker", | 
 |     "../../rtc_base:stringutils", | 
 |     "../../system_wrappers", | 
 |     "../../system_wrappers:field_trial_api", | 
 |     "../../system_wrappers:metrics_api", | 
 |     "../audio_coding:audio_format_conversion", | 
 |     "../remote_bitrate_estimator", | 
 |   ] | 
 |  | 
 |   # TODO(jschuh): Bug 1348: fix this warning. | 
 |   configs += [ "//build/config/compiler:no_size_t_to_int_warning" ] | 
 | } | 
 |  | 
 | rtc_source_set("rtcp_transceiver") { | 
 |   visibility = [ "*" ] | 
 |   public = [ | 
 |     "source/rtcp_transceiver.h", | 
 |     "source/rtcp_transceiver_config.h", | 
 |     "source/rtcp_transceiver_impl.h", | 
 |   ] | 
 |   sources = [ | 
 |     "source/rtcp_transceiver.cc", | 
 |     "source/rtcp_transceiver_config.cc", | 
 |     "source/rtcp_transceiver_impl.cc", | 
 |   ] | 
 |   deps = [ | 
 |     ":rtp_rtcp", | 
 |     ":rtp_rtcp_format", | 
 |     "../../:webrtc_common", | 
 |     "../../api:array_view", | 
 |     "../../api:optional", | 
 |     "../../api:transport_api", | 
 |     "../../rtc_base:checks", | 
 |     "../../rtc_base:rtc_base_approved", | 
 |     "../../rtc_base:rtc_task_queue", | 
 |     "../../rtc_base:weak_ptr", | 
 |     "../../system_wrappers", | 
 |   ] | 
 | } | 
 |  | 
 | rtc_source_set("fec_test_helper") { | 
 |   testonly = true | 
 |   sources = [ | 
 |     "source/fec_test_helper.cc", | 
 |     "source/fec_test_helper.h", | 
 |   ] | 
 |   deps = [ | 
 |     ":rtp_rtcp", | 
 |     ":rtp_rtcp_format", | 
 |     "..:module_api", | 
 |     "../../rtc_base:checks", | 
 |     "../../rtc_base:rtc_base_approved", | 
 |   ] | 
 |  | 
 |   # 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" ] | 
 |   } | 
 | } | 
 |  | 
 | rtc_source_set("mock_rtp_rtcp") { | 
 |   testonly = true | 
 |   sources = [ | 
 |     "mocks/mock_recovered_packet_receiver.h", | 
 |     "mocks/mock_rtcp_rtt_stats.h", | 
 |     "mocks/mock_rtp_rtcp.h", | 
 |   ] | 
 |   deps = [ | 
 |     ":rtp_rtcp", | 
 |     ":rtp_rtcp_format", | 
 |     "..:module_api", | 
 |     "../../api:optional", | 
 |     "../../rtc_base:checks", | 
 |     "../../rtc_base:rtc_base_approved", | 
 |     "../../test:test_support", | 
 |   ] | 
 | } | 
 |  | 
 | if (rtc_include_tests) { | 
 |   rtc_executable("test_packet_masks_metrics") { | 
 |     testonly = true | 
 |  | 
 |     sources = [ | 
 |       "test/testFec/average_residual_loss_xor_codes.h", | 
 |       "test/testFec/test_packet_masks_metrics.cc", | 
 |     ] | 
 |  | 
 |     deps = [ | 
 |       ":rtp_rtcp", | 
 |       "../../test:test_main", | 
 |       "//testing/gtest", | 
 |     ] | 
 |   }  # test_packet_masks_metrics | 
 |  | 
 |   rtc_source_set("rtp_rtcp_modules_tests") { | 
 |     testonly = true | 
 |  | 
 |     sources = [ | 
 |       "test/testFec/test_fec.cc", | 
 |     ] | 
 |     deps = [ | 
 |       ":rtp_rtcp", | 
 |       ":rtp_rtcp_format", | 
 |       "../../rtc_base:rtc_base_approved", | 
 |       "../../test:test_support", | 
 |     ] | 
 |     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" ] | 
 |     } | 
 |   } | 
 |  | 
 |   rtc_source_set("rtp_rtcp_unittests") { | 
 |     testonly = true | 
 |  | 
 |     sources = [ | 
 |       "source/byte_io_unittest.cc", | 
 |       "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_nack_stats_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/rrtr_unittest.cc", | 
 |       "source/rtcp_packet/sdes_unittest.cc", | 
 |       "source/rtcp_packet/sender_report_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_transceiver_impl_unittest.cc", | 
 |       "source/rtcp_transceiver_unittest.cc", | 
 |       "source/rtp_fec_unittest.cc", | 
 |       "source/rtp_format_h264_unittest.cc", | 
 |       "source/rtp_format_video_generic_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_map_unittest.cc", | 
 |       "source/rtp_packet_history_unittest.cc", | 
 |       "source/rtp_packet_unittest.cc", | 
 |       "source/rtp_payload_registry_unittest.cc", | 
 |       "source/rtp_receiver_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", | 
 |       "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 = [ | 
 |       ":fec_test_helper", | 
 |       ":mock_rtp_rtcp", | 
 |       ":rtcp_transceiver", | 
 |       ":rtp_rtcp", | 
 |       ":rtp_rtcp_format", | 
 |       "..:module_api", | 
 |       "../..:webrtc_common", | 
 |       "../../:typedefs", | 
 |       "../../api:array_view", | 
 |       "../../api:libjingle_peerconnection_api", | 
 |       "../../api:optional", | 
 |       "../../api:transport_api", | 
 |       "../../api:video_frame_api", | 
 |       "../../call:rtp_receiver", | 
 |       "../../common_video:common_video", | 
 |       "../../logging:mocks", | 
 |       "../../logging:rtc_event_log_api", | 
 |       "../../rtc_base:checks", | 
 |       "../../rtc_base:rate_limiter", | 
 |       "../../rtc_base:rtc_base_approved", | 
 |       "../../rtc_base:rtc_base_tests_utils", | 
 |       "../../rtc_base:rtc_task_queue", | 
 |       "../../system_wrappers", | 
 |       "../../test:field_trial", | 
 |       "../../test:rtp_test_utils", | 
 |       "../../test:test_common", | 
 |       "../../test:test_support", | 
 |       "../audio_coding:audio_format_conversion", | 
 |     ] | 
 |  | 
 |     # 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" ] | 
 |     } | 
 |   } | 
 | } |