Revert of Enable GN check for webrtc/base (patchset #9 id:350001 of https://codereview.webrtc.org/2840453004/ )

Reason for revert:
It causes a Chromium build error:

ERROR at //third_party/webrtc/test/BUILD.gn:113:5: Can't load input file.
    "//third_party/gflags",

Original issue's description:
> Reland of Enable GN check for webrtc/base (patchset #3 id:230001 of https://codereview.webrtc.org/2838683002/ )
>
> Reason for revert:
> Try to fix the webrtc/test/fuzzers issue and reland this CL because it
> contains lots of fixes for our BUILD.gn files.
>
> Original issue's description:
> > Revert of Enable GN check for webrtc/base (patchset #13 id:240001 of https://codereview.webrtc.org/2717083002/ )
> >
> > Reason for revert:
> > Breaks Chromium because in Chromium we import WebRTC with rtc_include_tests=false (https://bugs.chromium.org/p/chromium/issues/detail?id=713179#c6).
> >
> > Chromium uses webrtc/test/fuzzers and this CL adds test dependencies to neteq_rtc_fuzzer.
> >
> > Original issue's description:
> > > Enable GN check for webrtc/base
> > >
> > > It's not possible to enable it for the rtc_base_approved
> > > target but since a larger refactoring is ongoing for webrtc/base
> > > this CL doesn't attempt to fix that.
> > >
> > > Changes made:
> > > * Move webrtc/system_wrappers/include/stringize_macros.h into
> > >   webrtc/base:rtc_base_approved_unittests (and corresponding
> > >   unit test to rtc_base_approved_unittests).
> > > * Move md5digest.* from rtc_base_approved to rtc_base_test_utils target.
> > > * Move webrtc/system_wrappers/include/stringize_macros.h (+test) into
> > >   webrtc/base.
> > > * Remove unused use include of webrtc/base/fileutils.h in
> > >   webrtc/base/pathutils.cc
> > >
> > > BUG=webrtc:6828, webrtc:3806, webrtc:7480
> > > NOTRY=True
> > >
> > > Review-Url: https://codereview.webrtc.org/2717083002
> > > Cr-Commit-Position: refs/heads/master@{#17766}
> > > Committed: ed754e71ae
> >
> > TBR=perkj@webrtc.org,tommi@webrtc.org,nisse@webrtc.org,kjellander@webrtc.org
> > # Not skipping CQ checks because original CL landed more than 1 days ago.
> > BUG=webrtc:6828, webrtc:3806, webrtc:7480
> > NOTRY=True
> >
> > Review-Url: https://codereview.webrtc.org/2838683002
> > Cr-Commit-Position: refs/heads/master@{#17849}
> > Committed: 11ed366c48
>
> TBR=perkj@webrtc.org,tommi@webrtc.org,nisse@webrtc.org,kjellander@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6828, webrtc:3806, webrtc:7480
>
> Review-Url: https://codereview.webrtc.org/2840453004
> Cr-Commit-Position: refs/heads/master@{#17876}
> Committed: 7054085e59

TBR=perkj@webrtc.org,tommi@webrtc.org,nisse@webrtc.org,kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6828, webrtc:3806, webrtc:7480

Review-Url: https://codereview.webrtc.org/2846483002
Cr-Commit-Position: refs/heads/master@{#17877}
This commit is contained in:
mbonadei 2017-04-26 00:38:48 -07:00 committed by Commit bot
parent 7054085e59
commit 3d7b0e2fda
11 changed files with 167 additions and 194 deletions

1
.gn
View File

@ -24,7 +24,6 @@ secondary_source = "//build/secondary/"
check_targets = [
"//webrtc/api/*",
"//webrtc/audio/*",
"//webrtc/base/*",
"//webrtc/call/*",
"//webrtc/common_video/*",
"//webrtc/common_audio/*",

View File

@ -92,10 +92,6 @@ source_set("protobuf_utils") {
# The subset of rtc_base approved for use outside of libjingle.
rtc_static_library("rtc_base_approved") {
# TODO(kjellander): Remove (bugs.webrtc.org/7480)
# Enabling GN check triggers a cyclic dependency caused by rate_limiter.cc:
# :rtc_base_approved -> //webrtc/system_wrappers -> :rtc_base_approved
check_includes = false
defines = []
libs = []
deps = []
@ -140,6 +136,8 @@ rtc_static_library("rtc_base_approved") {
"location.h",
"md5.cc",
"md5.h",
"md5digest.cc",
"md5digest.h",
"mod_ops.h",
"onetimeevent.h",
"optional.cc",
@ -174,7 +172,6 @@ rtc_static_library("rtc_base_approved") {
"string_to_number.h",
"stringencode.cc",
"stringencode.h",
"stringize_macros.h",
"stringutils.cc",
"stringutils.h",
"swap_queue.h",
@ -658,74 +655,21 @@ rtc_source_set("gtest_prod") {
]
}
config("rtc_base_tests_utils_exported_config") {
defines = [ "GTEST_RELATIVE_PATH" ]
}
config("rtc_base_tests_utils_warnings_config") {
if (is_win && is_clang) {
cflags = [
# See https://bugs.chromium.org/p/webrtc/issues/detail?id=6270
"-Wno-reorder",
"-Wno-sign-compare",
]
}
}
rtc_source_set("rtc_base_tests_utils") {
testonly = true
sources = [
# Also use this as a convenient dumping ground for misc files that are
# included by multiple targets below.
"cpu_time.cc",
"cpu_time.h",
"fakeclock.cc",
"fakeclock.h",
"fakenetwork.h",
"fakesslidentity.h",
"firewallsocketserver.cc",
"firewallsocketserver.h",
"gunit.h",
"httpserver.cc",
"httpserver.h",
"md5digest.cc",
"md5digest.h",
"memory_usage.cc",
"memory_usage.h",
"natserver.cc",
"natserver.h",
"natsocketfactory.cc",
"natsocketfactory.h",
"nattypes.cc",
"nattypes.h",
"proxyserver.cc",
"proxyserver.h",
"sigslottester.h",
"sigslottester.h.pump",
"testbase64.h",
"testechoserver.h",
"testutils.h",
"timedelta.h",
]
configs += [ ":rtc_base_tests_utils_warnings_config" ]
public_configs = [ ":rtc_base_tests_utils_exported_config" ]
deps = [
":rtc_base",
"../test:field_trial",
"../test:test_support",
]
public_deps = [
"//testing/gmock",
"//testing/gtest",
]
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 (rtc_include_tests) {
config("rtc_base_tests_utils_exported_config") {
defines = [ "GTEST_RELATIVE_PATH" ]
}
config("rtc_base_tests_utils_warnings_config") {
if (is_win && is_clang) {
cflags = [
# See https://bugs.chromium.org/p/webrtc/issues/detail?id=6270
"-Wno-reorder",
"-Wno-sign-compare",
]
}
}
rtc_source_set("rtc_base_tests_main") {
testonly = true
sources = [
@ -733,9 +677,56 @@ if (rtc_include_tests) {
]
public_configs = [ ":rtc_base_tests_utils_exported_config" ]
deps = [
":rtc_base",
":rtc_base_approved",
":rtc_base_tests_utils",
]
public_deps = [
"//testing/gmock",
"//testing/gtest",
]
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("rtc_base_tests_utils") {
testonly = true
sources = [
# Also use this as a convenient dumping ground for misc files that are
# included by multiple targets below.
"cpu_time.cc",
"cpu_time.h",
"fakeclock.cc",
"fakeclock.h",
"fakenetwork.h",
"fakesslidentity.h",
"firewallsocketserver.cc",
"firewallsocketserver.h",
"gunit.h",
"httpserver.cc",
"httpserver.h",
"memory_usage.cc",
"memory_usage.h",
"natserver.cc",
"natserver.h",
"natsocketfactory.cc",
"natsocketfactory.h",
"nattypes.cc",
"nattypes.h",
"proxyserver.cc",
"proxyserver.h",
"sigslottester.h",
"sigslottester.h.pump",
"testbase64.h",
"testechoserver.h",
"testutils.h",
"timedelta.h",
]
configs += [ ":rtc_base_tests_utils_warnings_config" ]
public_configs = [ ":rtc_base_tests_utils_exported_config" ]
deps = [
":rtc_base",
"../test:field_trial",
"../test:test_support",
]
@ -772,9 +763,6 @@ if (rtc_include_tests) {
deps = [
":rtc_base",
":rtc_base_tests_main",
":rtc_base_tests_utils",
"../system_wrappers:system_wrappers",
"../test:test_support",
"//testing/gtest",
]
if (is_win) {
@ -829,7 +817,6 @@ if (rtc_include_tests) {
"safe_minmax_unittest.cc",
"string_to_number_unittest.cc",
"stringencode_unittest.cc",
"stringize_macros_unittest.cc",
"stringutils_unittest.cc",
"swap_queue_unittest.cc",
"thread_annotations_unittest.cc",
@ -838,13 +825,8 @@ if (rtc_include_tests) {
"timeutils_unittest.cc",
]
deps = [
":rtc_base",
":rtc_base_approved",
":rtc_base_tests_main",
":rtc_base_tests_utils",
":rtc_task_queue",
"../system_wrappers:system_wrappers",
"../test:test_support",
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
@ -868,9 +850,7 @@ if (rtc_include_tests) {
]
deps = [
":rtc_base_tests_main",
":rtc_base_tests_utils",
":rtc_task_queue",
"../test:test_support",
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
@ -892,10 +872,8 @@ if (rtc_include_tests) {
"numerics/percentile_filter_unittest.cc",
]
deps = [
":rtc_base_approved",
":rtc_base_tests_main",
":rtc_numerics",
"../test:test_support",
]
}
@ -961,8 +939,6 @@ if (rtc_include_tests) {
}
deps = [
":rtc_base_tests_main",
":rtc_base_tests_utils",
"../test:test_support",
]
public_deps = [
":rtc_base",

View File

@ -13,7 +13,7 @@
#include <string>
#include "webrtc/base/stringize_macros.h"
#include "webrtc/system_wrappers/include/stringize_macros.h"
namespace rtc {

View File

@ -16,6 +16,7 @@
#endif // WEBRTC_WIN
#include "webrtc/base/checks.h"
#include "webrtc/base/fileutils.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/pathutils.h"
#include "webrtc/base/stringutils.h"

View File

@ -19,11 +19,11 @@
#include <algorithm>
#include <memory>
#include "webrtc/base/stringize_macros.h"
#include "webrtc/base/timeutils.h"
#include "webrtc/common_audio/resampler/sinc_resampler.h"
#include "webrtc/common_audio/resampler/sinusoidal_linear_chirp_source.h"
#include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
#include "webrtc/system_wrappers/include/stringize_macros.h"
#include "webrtc/test/gmock.h"
#include "webrtc/test/gtest.h"

View File

@ -1130,101 +1130,6 @@ rtc_source_set("neteq_test_minimal") {
]
}
config("neteq_unittest_tools_config") {
include_dirs = [ "tools" ]
}
rtc_source_set("neteq_unittest_tools") {
testonly = true
sources = [
"neteq/tools/audio_checksum.h",
"neteq/tools/audio_loop.cc",
"neteq/tools/audio_loop.h",
"neteq/tools/audio_sink.cc",
"neteq/tools/audio_sink.h",
"neteq/tools/constant_pcm_packet_source.cc",
"neteq/tools/constant_pcm_packet_source.h",
"neteq/tools/fake_decode_from_file.cc",
"neteq/tools/fake_decode_from_file.h",
"neteq/tools/input_audio_file.cc",
"neteq/tools/input_audio_file.h",
"neteq/tools/neteq_input.h",
"neteq/tools/neteq_replacement_input.cc",
"neteq/tools/neteq_replacement_input.h",
"neteq/tools/output_audio_file.h",
"neteq/tools/output_wav_file.h",
"neteq/tools/packet.cc",
"neteq/tools/packet.h",
"neteq/tools/packet_source.cc",
"neteq/tools/packet_source.h",
"neteq/tools/resample_input_audio_file.cc",
"neteq/tools/resample_input_audio_file.h",
"neteq/tools/rtp_file_source.cc",
"neteq/tools/rtp_file_source.h",
"neteq/tools/rtp_generator.cc",
"neteq/tools/rtp_generator.h",
]
public_configs = [ ":neteq_unittest_tools_config" ]
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 = [
":audio_encoder_interface",
":pcm16b",
"../..:webrtc_common",
"../../api/audio_codecs:audio_codecs_api",
"../../base:rtc_base_approved",
"../../base:rtc_base_tests_utils",
"../../common_audio",
"../../test:rtp_test_utils",
"../rtp_rtcp",
]
public_deps = [
":neteq_test_minimal",
]
if (rtc_enable_protobuf) {
sources += [
"neteq/tools/neteq_packet_source_input.cc",
"neteq/tools/neteq_packet_source_input.h",
]
deps += [ ":rtc_event_log_source" ]
}
}
if (rtc_enable_protobuf) {
rtc_static_library("rtc_event_log_source") {
testonly = true
# TODO(kjellander): Remove (bugs.webrtc.org/6828)
# Needs call.h to be moved to webrtc/api first.
check_includes = false
sources = [
"neteq/tools/rtc_event_log_source.cc",
"neteq/tools/rtc_event_log_source.h",
]
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 = [
"../../base:rtc_base_approved",
"../../logging:rtc_event_log_parser",
]
public_deps = [
"../../logging:rtc_event_log_proto",
]
}
}
if (rtc_include_tests) {
group("audio_coding_tests") {
testonly = true
@ -1495,6 +1400,32 @@ if (rtc_include_tests) {
proto_out_dir = "webrtc/modules/audio_coding/neteq"
}
rtc_static_library("rtc_event_log_source") {
testonly = true
# TODO(kjellander): Remove (bugs.webrtc.org/6828)
# Needs call.h to be moved to webrtc/api first.
check_includes = false
sources = [
"neteq/tools/rtc_event_log_source.cc",
"neteq/tools/rtc_event_log_source.h",
]
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 = [
"../../base:rtc_base_approved",
"../../logging:rtc_event_log_parser",
]
public_deps = [
"../../logging:rtc_event_log_proto",
]
}
rtc_test("neteq_rtpplay") {
testonly = true
defines = []
@ -1612,6 +1543,72 @@ if (rtc_include_tests) {
]
}
config("neteq_unittest_tools_config") {
include_dirs = [ "tools" ]
}
rtc_source_set("neteq_unittest_tools") {
testonly = true
sources = [
"neteq/tools/audio_checksum.h",
"neteq/tools/audio_loop.cc",
"neteq/tools/audio_loop.h",
"neteq/tools/audio_sink.cc",
"neteq/tools/audio_sink.h",
"neteq/tools/constant_pcm_packet_source.cc",
"neteq/tools/constant_pcm_packet_source.h",
"neteq/tools/fake_decode_from_file.cc",
"neteq/tools/fake_decode_from_file.h",
"neteq/tools/input_audio_file.cc",
"neteq/tools/input_audio_file.h",
"neteq/tools/neteq_input.h",
"neteq/tools/neteq_replacement_input.cc",
"neteq/tools/neteq_replacement_input.h",
"neteq/tools/output_audio_file.h",
"neteq/tools/output_wav_file.h",
"neteq/tools/packet.cc",
"neteq/tools/packet.h",
"neteq/tools/packet_source.cc",
"neteq/tools/packet_source.h",
"neteq/tools/resample_input_audio_file.cc",
"neteq/tools/resample_input_audio_file.h",
"neteq/tools/rtp_file_source.cc",
"neteq/tools/rtp_file_source.h",
"neteq/tools/rtp_generator.cc",
"neteq/tools/rtp_generator.h",
]
public_configs = [ ":neteq_unittest_tools_config" ]
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 = [
":audio_encoder_interface",
":pcm16b",
"../..:webrtc_common",
"../../api/audio_codecs:audio_codecs_api",
"../../base:rtc_base_approved",
"../../common_audio",
"../../test:rtp_test_utils",
"../rtp_rtcp",
]
public_deps = [
":neteq_test_minimal",
]
if (rtc_enable_protobuf) {
sources += [
"neteq/tools/neteq_packet_source_input.cc",
"neteq/tools/neteq_packet_source_input.h",
]
deps += [ ":rtc_event_log_source" ]
}
}
rtc_source_set("neteq_test_tools") {
testonly = true
sources = [

View File

@ -27,6 +27,7 @@ rtc_static_library("system_wrappers") {
"include/rw_lock_wrapper.h",
"include/sleep.h",
"include/static_instance.h",
"include/stringize_macros.h",
"include/timestamp_extrapolator.h",
"include/trace.h",
"source/aligned_malloc.cc",
@ -210,6 +211,7 @@ if (rtc_include_tests) {
"source/metrics_unittest.cc",
"source/ntp_time_unittest.cc",
"source/rtp_to_ntp_estimator_unittest.cc",
"source/stringize_macros_unittest.cc",
]
configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]

View File

@ -15,8 +15,8 @@
// symbols (or their output) and manipulating preprocessor symbols
// that define strings.
#ifndef WEBRTC_BASE_STRINGIZE_MACROS_H_
#define WEBRTC_BASE_STRINGIZE_MACROS_H_
#ifndef WEBRTC_SYSTEM_WRAPPERS_INCLUDE_STRINGIZE_MACROS_H_
#define WEBRTC_SYSTEM_WRAPPERS_INCLUDE_STRINGIZE_MACROS_H_
// This is not very useful as it does not expand defined symbols if
// called directly. Use its counterpart without the _NO_EXPANSION
@ -35,4 +35,4 @@
// STRINGIZE(B(y)) produces "myobj->FunctionCall(y)"
#define STRINGIZE(x) STRINGIZE_NO_EXPANSION(x)
#endif // WEBRTC_BASE_STRINGIZE_MACROS_H_
#endif // WEBRTC_SYSTEM_WRAPPERS_INCLUDE_STRINGIZE_MACROS_H_

View File

@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/base/stringize_macros.h"
#include "webrtc/system_wrappers/include/stringize_macros.h"
#include "webrtc/test/gtest.h"

View File

@ -257,9 +257,8 @@ webrtc_fuzzer_test("neteq_rtp_fuzzer") {
"neteq_rtp_fuzzer.cc",
]
deps = [
"../../base:rtc_base_approved",
"../../modules/audio_coding:neteq",
"../../modules/audio_coding:neteq_test_minimal",
"../../modules/audio_coding:neteq_unittest_tools",
"../../modules/audio_coding:pcm16b",
"../../modules/rtp_rtcp",
]

View File

@ -181,7 +181,6 @@ if (rtc_include_tests) {
":file_player",
":voice_engine",
"../base:rtc_base_approved",
"../base:rtc_base_tests_utils",
"../test:test_common",
"//testing/gmock",
"//testing/gtest",