Fixing WebRTC after moving from src/webrtc to src/
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}diff --git a/PRESUBMIT.py b/PRESUBMIT.py
index 927869f..66d6892 100755
--- a/PRESUBMIT.py
+++ b/PRESUBMIT.py
@@ -15,29 +15,29 @@
# Files and directories that are *skipped* by cpplint in the presubmit script.
CPPLINT_BLACKLIST = [
+ 'api/video_codecs/video_decoder.h',
+ 'common_types.cc',
+ 'common_types.h',
+ 'examples/objc',
+ 'media',
+ 'modules/audio_coding',
+ 'modules/audio_conference_mixer',
+ 'modules/audio_device',
+ 'modules/audio_processing',
+ 'modules/desktop_capture',
+ 'modules/include/module_common_types.h',
+ 'modules/media_file',
+ 'modules/utility',
+ 'modules/video_capture',
+ 'p2p',
+ 'pc',
+ 'rtc_base',
+ 'sdk/android/src/jni',
+ 'sdk/objc',
+ 'system_wrappers',
+ 'test',
'tools_webrtc',
- 'webrtc/api/video_codecs/video_decoder.h',
- 'webrtc/examples/objc',
- 'webrtc/media',
- 'webrtc/modules/audio_coding',
- 'webrtc/modules/audio_conference_mixer',
- 'webrtc/modules/audio_device',
- 'webrtc/modules/audio_processing',
- 'webrtc/modules/desktop_capture',
- 'webrtc/modules/include/module_common_types.h',
- 'webrtc/modules/media_file',
- 'webrtc/modules/utility',
- 'webrtc/modules/video_capture',
- 'webrtc/p2p',
- 'webrtc/pc',
- 'webrtc/rtc_base',
- 'webrtc/sdk/android/src/jni',
- 'webrtc/sdk/objc',
- 'webrtc/system_wrappers',
- 'webrtc/test',
- 'webrtc/voice_engine',
- 'webrtc/common_types.h',
- 'webrtc/common_types.cc',
+ 'voice_engine',
]
# These filters will always be removed, even if the caller specifies a filter
@@ -62,34 +62,33 @@
# webrtc-users@google.com (internal list).
# 4. (later) The deprecated APIs are removed.
NATIVE_API_DIRS = (
- 'webrtc',
- 'webrtc/api',
- 'webrtc/media',
- 'webrtc/modules/audio_device/include',
- 'webrtc/pc',
+ 'api',
+ 'media',
+ 'modules/audio_device/include',
+ 'pc',
)
# These directories should not be used but are maintained only to avoid breaking
# some legacy downstream code.
LEGACY_API_DIRS = (
- 'webrtc/common_audio/include',
- 'webrtc/modules/audio_coding/include',
- 'webrtc/modules/audio_conference_mixer/include',
- 'webrtc/modules/audio_processing/include',
- 'webrtc/modules/bitrate_controller/include',
- 'webrtc/modules/congestion_controller/include',
- 'webrtc/modules/include',
- 'webrtc/modules/remote_bitrate_estimator/include',
- 'webrtc/modules/rtp_rtcp/include',
- 'webrtc/modules/rtp_rtcp/source',
- 'webrtc/modules/utility/include',
- 'webrtc/modules/video_coding/codecs/h264/include',
- 'webrtc/modules/video_coding/codecs/i420/include',
- 'webrtc/modules/video_coding/codecs/vp8/include',
- 'webrtc/modules/video_coding/codecs/vp9/include',
- 'webrtc/modules/video_coding/include',
- 'webrtc/rtc_base',
- 'webrtc/system_wrappers/include',
- 'webrtc/voice_engine/include',
+ 'common_audio/include',
+ 'modules/audio_coding/include',
+ 'modules/audio_conference_mixer/include',
+ 'modules/audio_processing/include',
+ 'modules/bitrate_controller/include',
+ 'modules/congestion_controller/include',
+ 'modules/include',
+ 'modules/remote_bitrate_estimator/include',
+ 'modules/rtp_rtcp/include',
+ 'modules/rtp_rtcp/source',
+ 'modules/utility/include',
+ 'modules/video_coding/codecs/h264/include',
+ 'modules/video_coding/codecs/i420/include',
+ 'modules/video_coding/codecs/vp8/include',
+ 'modules/video_coding/codecs/vp9/include',
+ 'modules/video_coding/include',
+ 'rtc_base',
+ 'system_wrappers/include',
+ 'voice_engine/include',
)
API_DIRS = NATIVE_API_DIRS[:] + LEGACY_API_DIRS[:]
@@ -331,8 +330,7 @@
cwd = input_api.PresubmitLocalPath()
script_path = os.path.join('tools_webrtc', 'presubmit_checks_lib',
'check_package_boundaries.py')
- webrtc_path = os.path.join('webrtc')
- command = [sys.executable, script_path, webrtc_path]
+ command = [sys.executable, script_path]
command += [gn_file.LocalPath() for gn_file in gn_files]
returncode, _, stderr = _RunCommand(command, cwd)
if returncode:
@@ -347,8 +345,7 @@
gn_files = []
for f in input_api.AffectedSourceFiles(source_file_filter):
- if f.LocalPath().startswith('webrtc'):
- gn_files.append(f)
+ gn_files.append(f)
result = []
if gn_files:
@@ -494,9 +491,9 @@
test_directories = [
input_api.PresubmitLocalPath(),
- Join('webrtc', 'rtc_tools', 'py_event_log_analyzer'),
- Join('webrtc', 'rtc_tools'),
- Join('webrtc', 'audio', 'test', 'unittests'),
+ Join('rtc_tools', 'py_event_log_analyzer'),
+ Join('rtc_tools'),
+ Join('audio', 'test', 'unittests'),
] + [
root for root, _, files in os.walk(Join('tools_webrtc'))
if any(f.endswith('_test.py') for f in files)
@@ -517,7 +514,7 @@
"""Checks that the namespace google::protobuf has not been used."""
files = []
pattern = input_api.re.compile(r'google::protobuf')
- proto_utils_path = os.path.join('webrtc', 'rtc_base', 'protobuf_utils.h')
+ proto_utils_path = os.path.join('rtc_base', 'protobuf_utils.h')
for f in input_api.AffectedSourceFiles(input_api.FilterSourceFile):
if f.LocalPath() in [proto_utils_path, 'PRESUBMIT.py']:
continue
@@ -533,6 +530,28 @@
return []
+def _LicenseHeader(input_api):
+ """Returns the license header regexp."""
+ # Accept any year number from 2003 to the current year
+ current_year = int(input_api.time.strftime('%Y'))
+ allowed_years = (str(s) for s in reversed(xrange(2003, current_year + 1)))
+ years_re = '(' + '|'.join(allowed_years) + ')'
+ license_header = (
+ r'.*? Copyright( \(c\))? %(year)s The WebRTC [Pp]roject [Aa]uthors\. '
+ r'All [Rr]ights [Rr]eserved\.\n'
+ r'.*?\n'
+ r'.*? Use of this source code is governed by a BSD-style license\n'
+ r'.*? that can be found in the LICENSE file in the root of the source\n'
+ r'.*? tree\. An additional intellectual property rights grant can be '
+ r'found\n'
+ r'.*? in the file PATENTS\. All contributing project authors may\n'
+ r'.*? be found in the AUTHORS file in the root of the source tree\.\n'
+ ) % {
+ 'year': years_re,
+ }
+ return license_header
+
+
def CommonChecks(input_api, output_api):
"""Checks common to both upload and commit."""
results = []
@@ -541,11 +560,12 @@
black_list = input_api.DEFAULT_BLACK_LIST + (
r".*\bobjc[\\\/].*",
r".*objc\.[hcm]+$",
- r"webrtc\/build\/ios\/SDK\/.*",
)
source_file_filter = lambda x: input_api.FilterSourceFile(x, None, black_list)
results.extend(CheckApprovedFilesLintClean(
input_api, output_api, source_file_filter))
+ results.extend(input_api.canned_checks.CheckLicense(
+ input_api, output_api, _LicenseHeader(input_api)))
results.extend(input_api.canned_checks.RunPylint(input_api, output_api,
black_list=(r'^base[\\\/].*\.py$',
r'^build[\\\/].*\.py$',
@@ -599,8 +619,13 @@
results.extend(CheckJSONParseErrors(input_api, output_api))
results.extend(RunPythonTests(input_api, output_api))
results.extend(CheckUsageOfGoogleProtobufNamespace(input_api, output_api))
- results.extend(CheckOrphanHeaders(input_api, output_api))
- results.extend(CheckNewLineAtTheEndOfProtoFiles(input_api, output_api))
+ # TODO(mbonadei): re-enable after the migration from src/webrtc to src/
+ # in order to avoid to trigger an error for each orphan header (we are
+ # moving all of them).
+ # results.extend(CheckOrphanHeaders(input_api, output_api))
+ # TODO(mbonadei): check before re-enable because it seems it is reporting
+ # some false positives.
+ # results.extend(CheckNewLineAtTheEndOfProtoFiles(input_api, output_api))
return results