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Revert "Add mock_rtc_event_log.h."

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This reverts commit 63aea46a6e424c35f7df1ec6e702c5eeb704996a.

Reason for revert: 
Breaks Chromium build:
https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2Fios-simulator%2F6721%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files__mb_%2F0%2Fstdout

ERROR at //third_party/webrtc/logging/BUILD.gn:295:5: Can't load input file.
    "../../test:test_support",
    ^------------------------
Unable to load:
  /b/build/slave/mac64/build/src/third_party/test/BUILD.gn
I also checked in the secondary tree for:
  /b/build/slave/mac64/build/src/build/secondary/third_party/test/BUILD.gn

Maybe this should be guarded by an "if (!build_with_chromium)"?

Original change's description:
> Add mock_rtc_event_log.h.
> 
> Bug: webrtc:7642
> Change-Id: Id3aa84d79e5e1a0520a968117cee550c9dd33c16
> Reviewed-on: https://webrtc-review.googlesource.com/37040
> Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21475}

TBR=phoglund@webrtc.org,mbonadei@webrtc.org

Change-Id: Ib49c7812261c46226ec0b7b3c99af2c3a8b78add
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7642
Reviewed-on: https://webrtc-review.googlesource.com/36981
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21477}
This commit is contained in:
Edward Lemur 2018-01-02 18:26:40 +00:00 committed by Commit Bot
parent 49acb1e8ca
commit 5a25ab298d
7 changed files with 0 additions and 18 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

1
audio/BUILD.gn
View File

@ -105,7 +105,6 @@ if (rtc_include_tests) {
"../call:mock_rtp_interfaces",
"../call:rtp_interfaces",
"../call:rtp_receiver",
"../logging:mocks",
"../modules/audio_device:mock_audio_device",
"../modules/audio_mixer:audio_mixer_impl",
"../modules/audio_processing:audio_processing_statistics",

12
logging/BUILD.gn
View File

@ -235,7 +235,6 @@ if (rtc_enable_protobuf) {
}
}
}
if (rtc_include_tests) {
rtc_executable("rtc_event_log2text") {
testonly = true
@ -284,14 +283,3 @@ if (rtc_enable_protobuf) {
}
}
}
rtc_source_set("mocks") {
testonly = true
sources = [
"rtc_event_log/mock/mock_rtc_event_log.h",
]
deps = [
":rtc_event_log_api",
"../../test:test_support",
]
}

1
modules/audio_coding/BUILD.gn
View File

@ -2150,7 +2150,6 @@ if (rtc_include_tests) {
"../../api/audio_codecs/opus:audio_encoder_opus",
"../../common_audio",
"../../common_audio:mock_common_audio",
"../../logging:mocks",
"../../logging:rtc_event_log_api",
"../../rtc_base:checks",
"../../rtc_base:protobuf_utils",

1
modules/bitrate_controller/BUILD.gn
View File

@ -64,7 +64,6 @@ if (rtc_include_tests) {
]
deps = [
":bitrate_controller",
"../../logging:mocks",
"../../logging:rtc_event_log_api",
"../../test:field_trial",
"../../test:test_support",

1
modules/congestion_controller/BUILD.gn
View File

@ -90,7 +90,6 @@ if (rtc_include_tests) {
deps = [
":congestion_controller",
":mock_congestion_controller",
"../../logging:mocks",
"../../rtc_base:checks",
"../../rtc_base:rtc_base",
"../../rtc_base:rtc_base_approved",

1
modules/remote_bitrate_estimator/BUILD.gn
View File

@ -143,7 +143,6 @@ if (rtc_include_tests) {
"../..:webrtc_common",
"../../:typedefs",
"../../api:optional",
"../../logging:mocks",
"../../rtc_base:checks",
"../../rtc_base:gtest_prod",
"../../rtc_base:rtc_base",

1
modules/rtp_rtcp/BUILD.gn
View File

@ -412,7 +412,6 @@ if (rtc_include_tests) {
"../../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",