35ac983a7e
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1366 4adac7df-926f-26a2-2b94-8c16560cd09d
112 lines
3.5 KiB
Plaintext
112 lines
3.5 KiB
Plaintext
# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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#
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# Use of this source code is governed by a BSD-style license
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# that can be found in the LICENSE file in the root of the source
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# tree. An additional intellectual property rights grant can be found
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# in the file PATENTS. All contributing project authors may
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# be found in the AUTHORS file in the root of the source tree.
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# Inspired by chromium.org:
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# http://dev.chromium.org/developers/contributing-code/watchlists
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{
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'WATCHLIST_DEFINITIONS': {
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'this_file': {
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'filepath': '^WATCHLISTS$',
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},
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'all_src': {
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'filepath': 'src/.*',
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},
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'root_files': {
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# src/build/ and non-recursive contents of ./ and src/
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'filepath': '^[^/]*$|src/[^/]*$|src/build/.*',
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},
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'documented_interfaces': {
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'filepath': 'src/[^/]*\.h$|'\
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'src/video_engine/main/interface/.*|'\
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'src/voice_engine/main/interface/.*',
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},
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'build_files': {
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'filepath': '\.gyp$|\.gypi$|Android\.mk$',
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},
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'java_files': {
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'filepath': '\.java$|\.xml$',
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},
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'video_engine': {
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'filepath': 'src/video_engine/.*',
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},
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'voice_engine': {
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'filepath': 'src/voice_engine/.*',
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},
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'common_audio': {
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'filepath': 'src/common_audio/.*',
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},
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'video_capture': {
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'filepath': 'src/modules/video_capture/.*',
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},
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'video_render': {
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'filepath': 'src/modules/video_render/.*',
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},
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'audio_device': {
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'filepath': 'src/modules/audio_device/.*',
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},
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'audio_coding': {
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'filepath': 'src/modules/audio_coding/.*',
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},
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'NetEQ': {
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'filepath': 'src/modules/audio_coding/NetEQ/.*',
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},
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'audio_processing': {
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'filepath': 'src/modules/audio_processing/.*',
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},
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'video_codecs': {
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'filepath': 'src/modules/video_coding/codecs/.*',
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},
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'video_coding': {
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'filepath': 'src/modules/video_coding/.*',
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},
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'rtp_rtcp': {
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'filepath': 'src/modules/rtp_rtcp/.*'
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},
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'system_wrappers': {
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'filepath': 'src/system_wrappers/.*',
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},
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},
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'WATCHLISTS': {
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'this_file': [''],
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'all_src': ['tterriberry@mozilla.com',
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'giles@mozilla.com'],
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'root_files': ['andrew@webrtc.org',
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'niklas.enbom@webrtc.org'],
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'documented_interfaces': ['interface-changes@webrtc.org',
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'rwolff@gocast.it'],
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'build_files': ['leozwang@webrtc.org'],
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'java_files': ['leozwang@webrtc.org'],
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'common_audio': ['bjornv@webrtc.org',
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'andrew@webrtc.org'],
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'video_engine': ['mflodman@webrtc.org',
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'perkj@webrtc.org'],
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'voice_engine': ['henrika@webrtc.org'],
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'video_capture': ['mflodman@webrtc.org',
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'perkj@webrtc.org',
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'leozwang@webrtc.org'],
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'video_render': ['mflodman@webrtc.org',
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'perkj@webrtc.org',
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'leozwang@webrtc.org'],
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'audio_device': ['henrika@webrtc.org',
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'leozwang@webrtc.org'],
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'audio_coding': ['tina.legrand@webrtc.org'],
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'NetEQ': ['henrik.lundin@webrtc.org'],
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'audio_processing': ['andrew@webrtc.org',
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'bjornv@webrtc.org'],
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'video_codecs': ['henrik.lundin@webrtc.org'],
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'video_coding': ['stefan@webrtc.org'],
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'rtp_rtcp': ['mflodman@webrtc.org',
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'pwestin@webrtc.org'],
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'system_wrappers': ['mflodman@webrtc.org',
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'henrika@webrtc.org',
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'andrew@webrtc.org'],
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},
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}
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