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Do not build rtp_generator in Chromium builds.

Browse Source 
This tool is only needed for WebRTC standalone so there is no need to
build it on Chromium trybots (if we want to do that, we need to
explicitly link against the Chromium's TQ implementation).

[1] - https://cs.chromium.org/chromium/src/third_party/webrtc_overrides/BUILD.gn?l=114-124&rcl=3514203635d4f5c2d660784dd3007f1018c9af88

Bug: None
Change-Id: Ib75204205717637e6b9b4320deaad5221ce35692
Reviewed-on: https://webrtc-review.googlesource.com/c/121405
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26533}
This commit is contained in:
Mirko Bonadei 2019-02-04 15:21:46 +01:00 committed by Commit Bot
parent 80a8687082
commit 8573aae7d0
1 changed files with 44 additions and 44 deletions
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88
rtc_tools/BUILD.gn
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@ -137,50 +137,6 @@ rtc_executable("frame_analyzer") {
]
}
rtc_executable("rtp_generator") {
visibility = [ "*" ]
testonly = true
sources = [
"rtp_generator/main.cc",
"rtp_generator/rtp_generator.cc",
"rtp_generator/rtp_generator.h",
]
deps = [
":command_line_parser",
"../api:libjingle_peerconnection_api",
"../api:transport_api",
"../api/video:builtin_video_bitrate_allocator_factory",
"../api/video_codecs:builtin_video_decoder_factory",
"../api/video_codecs:builtin_video_encoder_factory",
"../api/video_codecs:video_codecs_api",
"../call",
"../call:call_interfaces",
"../call:fake_network",
"../call:rtp_interfaces",
"../call:rtp_sender",
"../call:simulated_network",
"../call:simulated_packet_receiver",
"../call:video_stream_api",
"../logging:rtc_event_log_api",
"../logging:rtc_event_log_impl_base",
"../media:rtc_audio_video",
"../media:rtc_media_base",
"../rtc_base:rtc_base",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_json",
"../rtc_base/system:file_wrapper",
"../test:fileutils",
"../test:rtp_test_utils",
"../test:video_test_common",
"//third_party/abseil-cpp/absl/strings",
]
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" ]
}
}
if (!build_with_chromium && !build_with_mozilla) {
action("frame_analyzer_host") {
script = "//tools_webrtc/executable_host_build.py"
@ -197,6 +153,50 @@ if (!build_with_chromium && !build_with_mozilla) {
# Only expose the targets needed by Chromium (e.g. frame_analyzer) to avoid
# building a lot of redundant code as part of Chromium builds.
if (!build_with_chromium) {
rtc_executable("rtp_generator") {
visibility = [ "*" ]
testonly = true
sources = [
"rtp_generator/main.cc",
"rtp_generator/rtp_generator.cc",
"rtp_generator/rtp_generator.h",
]
deps = [
":command_line_parser",
"../api:libjingle_peerconnection_api",
"../api:transport_api",
"../api/video:builtin_video_bitrate_allocator_factory",
"../api/video_codecs:builtin_video_decoder_factory",
"../api/video_codecs:builtin_video_encoder_factory",
"../api/video_codecs:video_codecs_api",
"../call",
"../call:call_interfaces",
"../call:fake_network",
"../call:rtp_interfaces",
"../call:rtp_sender",
"../call:simulated_network",
"../call:simulated_packet_receiver",
"../call:video_stream_api",
"../logging:rtc_event_log_api",
"../logging:rtc_event_log_impl_base",
"../media:rtc_audio_video",
"../media:rtc_media_base",
"../rtc_base:rtc_base",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_json",
"../rtc_base/system:file_wrapper",
"../test:fileutils",
"../test:rtp_test_utils",
"../test:video_test_common",
"//third_party/abseil-cpp/absl/strings",
]
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_executable("psnr_ssim_analyzer") {
sources = [
"psnr_ssim_analyzer/psnr_ssim_analyzer.cc",