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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 [email protected] Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <[email protected]> Reviewed-by: Henrik Kjellander <[email protected]> Commit-Queue: Mirko Bonadei <[email protected]> Cr-Commit-Position: refs/heads/master@{#19846}
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| Original file line number | Diff line number | Diff line change |
|---|---|---|
|
|
@@ -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 | ||
|
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@@ -62,34 +62,33 @@ | |
| # [email protected] (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[:] | ||
|
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||
|
|
@@ -331,8 +330,7 @@ def CheckNoPackageBoundaryViolations(input_api, gn_files, output_api): | |
| 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: | ||
|
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@@ -347,8 +345,7 @@ def CheckGnChanges(input_api, output_api): | |
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||
| 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: | ||
|
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@@ -494,9 +491,9 @@ def Join(*args): | |
|
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| 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 @@ def CheckUsageOfGoogleProtobufNamespace(input_api, output_api): | |
| """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 | ||
|
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@@ -533,6 +530,28 @@ def CheckUsageOfGoogleProtobufNamespace(input_api, output_api): | |
| return [] | ||
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|
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| 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 | ||
|
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||
|
|
||
| def CommonChecks(input_api, output_api): | ||
| """Checks common to both upload and commit.""" | ||
| results = [] | ||
|
|
@@ -541,11 +560,12 @@ def CommonChecks(input_api, output_api): | |
| 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 @@ def CommonChecks(input_api, output_api): | |
| 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 | ||
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