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Refactor webrtc/modules/rtp_rtcp for GN check

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This moves some GN check configurations out of .gn to individual
targets.

This commit also removes the source file 'mocks/mock_rtp_rtcp.h' from
the static_library 'rtp_rtcp' because it depends on a 'testonly = true'
target. After a check this seems only included in the unitest code:

$ grep -Rn "mocks/mock_rtp_rtcp.h" webrtc/modules/rtp_rtcp/
webrtc/modules/rtp_rtcp/source/ulpfec_receiver_unittest.cc:18:#include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc:17:#include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"

This commit also removes the dependency on
'//webrt/modules/video_coding' because it seems that the following
include can be removed:

#include "webrtc/modules/video_coding/include/video_coding_defines.h"

The now checked target is:
"//webrtc/modules/rtp_rtcp/*"

BUG=webrtc:6828
NOTRY=True

Review-Url: https://codereview.webrtc.org/2598963002
Cr-Commit-Position: refs/heads/master@{#15760}
This commit is contained in:
mbonadei 2016-12-22 07:35:39 -08:00 committed by Commit bot
parent 000d16396e
commit ebafdc8484
4 changed files with 7 additions and 2 deletions
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1
.gn
View File

@ -31,6 +31,7 @@ check_targets = [
"//webrtc/modules/audio_processing/*",
"//webrtc/modules/media_file/*",
"//webrtc/modules/pacing/*",
"//webrtc/modules/rtp_rtcp/*",
"//webrtc/modules/video_capture/*",
"//webrtc/modules/video_coding/*",
"//webrtc/stats:rtc_stats",

1
webrtc/modules/BUILD.gn
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@ -421,6 +421,7 @@ if (rtc_include_tests) {
"remote_bitrate_estimator/test/bwe_unittest.cc",
"remote_bitrate_estimator/test/estimators/nada_unittest.cc",
"remote_bitrate_estimator/test/metric_recorder_unittest.cc",
"rtp_rtcp/mocks/mock_rtp_rtcp.h",
"rtp_rtcp/source/byte_io_unittest.cc",
"rtp_rtcp/source/fec_test_helper.cc",
"rtp_rtcp/source/fec_test_helper.h",

6
webrtc/modules/rtp_rtcp/BUILD.gn
View File

@ -20,7 +20,6 @@ rtc_static_library("rtp_rtcp") {
"include/rtp_rtcp.h",
"include/rtp_rtcp_defines.h",
"include/ulpfec_receiver.h",
"mocks/mock_rtp_rtcp.h",
"source/byte_io.h",
"source/dtmf_queue.cc",
"source/dtmf_queue.h",
@ -175,7 +174,12 @@ rtc_static_library("rtp_rtcp") {
deps = [
"../..:webrtc_common",
"../../api:transport_api",
"../../base:gtest_prod",
"../../base:rtc_base_approved",
"../../base:rtc_task_queue",
"../../call:call_interfaces",
"../../common_video",
"../../logging:rtc_event_log_api",
"../../system_wrappers",
"../remote_bitrate_estimator",
]

1
webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
View File

@ -22,7 +22,6 @@
#include "webrtc/modules/include/module.h"
#include "webrtc/modules/rtp_rtcp/include/flexfec_sender.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/video_coding/include/video_coding_defines.h"
namespace webrtc {