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webrtc_m130/modules/rtp_rtcp/BUILD.gn
Per Kjellander a7f2d84ad1 Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" 
This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613.

Reason for revert: 
The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660

Original change's description:
> Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""
> 
> This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16.
> 
> Reason for revert: 
> 
> Breaks Chrome FYI:
> 
> /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check
>   -> returned 1
> ERROR at //build/split_static_library.gni:12:5: Dependency not allowed.
>     static_library(target_name) {
>     ^----------------------------
> The item //content/renderer:renderer
> can not depend on //third_party/webrtc/media:rtc_internal_video_codecs
> because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [
>   //third_party/webrtc/*
>   //third_party/webrtc_overrides/*
> ]
> 
>  https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout
> 
> Original change's description:
> > GN rtc_* templates: Set default visibility to webrtc_root + "/*"
> > 
> > This means that by default, targets are visible to everything under
> > the WebRTC root, but not visible to anything else.
> > 
> > API targets are manually tagged with visibility "*", so that targets
> > outside the WebRTC tree can see them.
> > 
> > BUG=webrtc:8254
> > 
> > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509
> > Reviewed-on: https://webrtc-review.googlesource.com/24140
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21548}
> 
> TBR=mbonadei@webrtc.org,kwiberg@webrtc.org
> 
> Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8254
> Reviewed-on: https://webrtc-review.googlesource.com/38760
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21555}

TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org

Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8254
Reviewed-on: https://webrtc-review.googlesource.com/38860
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:55:04 +00:00

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# 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_video_stereo.cc",
"source/rtp_format_video_stereo.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_log_api",
"../../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" ]
if (is_win) {
cflags = [
# TODO(kjellander): Bug 261: fix this warning.
"/wd4373", # virtual function override.
]
}
}
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_video_stereo_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",
"//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" ]
}
}
}
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