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Stop using public_deps in the call module.

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Bug: webrtc:8603
Change-Id: I048127bc86f213e638e6814ac8a86761cb8a64db
Reviewed-on: https://webrtc-review.googlesource.com/28624
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21072}
This commit is contained in:
Mirko Bonadei
2017-12-04 10:50:51 +01:00
committed by Commit Bot
parent 3ffc03edad
commit a0e1a55dc9
4 changed files with 6 additions and 6 deletions
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7
call/BUILD.gn
View File

@ -121,12 +121,6 @@ rtc_static_library("call") {
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
} }
public_deps = [
":call_interfaces",
"../api:call_api",
"../api:libjingle_peerconnection_api",
]
deps = [ deps = [
":bitrate_allocator", ":bitrate_allocator",
":call_interfaces", ":call_interfaces",
@ -196,6 +190,7 @@ if (rtc_include_tests) {
deps = [ deps = [
":bitrate_allocator", ":bitrate_allocator",
":call", ":call",
":call_interfaces",
":mock_rtp_interfaces", ":mock_rtp_interfaces",
":rtp_interfaces", ":rtp_interfaces",
":rtp_receiver", ":rtp_receiver",

2
logging/BUILD.gn
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@ -177,7 +177,9 @@ if (rtc_enable_protobuf) {
deps = [ deps = [
":rtc_event_log_impl", ":rtc_event_log_impl",
":rtc_event_log_parser", ":rtc_event_log_parser",
"../api:libjingle_peerconnection_api",
"../call", "../call",
"../call:call_interfaces",
"../modules/audio_coding:audio_network_adaptor", "../modules/audio_coding:audio_network_adaptor",
"../modules/remote_bitrate_estimator:remote_bitrate_estimator", "../modules/remote_bitrate_estimator:remote_bitrate_estimator",
"../modules/rtp_rtcp", "../modules/rtp_rtcp",

1
media/BUILD.gn
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@ -128,6 +128,7 @@ rtc_static_library("rtc_audio_video") {
libs = [] libs = []
deps = [ deps = [
"../api:video_frame_api_i420", "../api:video_frame_api_i420",
"../call:call_interfaces",
"../modules/video_coding:video_coding_utility", "../modules/video_coding:video_coding_utility",
] ]
sources = [ sources = [

2
test/BUILD.gn
View File

@ -465,6 +465,7 @@ rtc_source_set("direct_transport") {
"..:webrtc_common", "..:webrtc_common",
"../api:transport_api", "../api:transport_api",
"../call", "../call",
"../call:call_interfaces",
"../modules/rtp_rtcp", "../modules/rtp_rtcp",
"../rtc_base:rtc_base_approved", "../rtc_base:rtc_base_approved",
"../rtc_base:sequenced_task_checker", "../rtc_base:sequenced_task_checker",
@ -563,6 +564,7 @@ rtc_source_set("test_common") {
"../api/video_codecs:video_codecs_api", "../api/video_codecs:video_codecs_api",
"../audio", "../audio",
"../call", "../call",
"../call:call_interfaces",
"../call:rtp_sender", "../call:rtp_sender",
"../call:video_stream_api", "../call:video_stream_api",
"../common_video", "../common_video",