Create gyp target for bwe components.
R=henrikg@webrtc.org, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1775004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4311 4adac7df-926f-26a2-2b94-8c16560cd09d
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@ -41,6 +41,7 @@
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'modules_java_gyp_path%': '<(modules_java_gyp_path)',
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'webrtc_vp8_dir%': '<(webrtc_root)/modules/video_coding/codecs/vp8',
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'rbe_components_path%': '<(webrtc_root)/modules/remote_bitrate_estimator',
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'include_opus%': 1,
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},
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'build_with_chromium%': '<(build_with_chromium)',
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@ -50,6 +51,7 @@
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'modules_java_gyp_path%': '<(modules_java_gyp_path)',
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'webrtc_vp8_dir%': '<(webrtc_vp8_dir)',
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'include_opus%': '<(include_opus)',
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'rbe_components_path%': '<(rbe_components_path)',
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# The Chromium common.gypi we use treats all gyp files without
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# chromium_code==1 as third party code. This disables many of the
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@ -87,6 +87,7 @@
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'webrtc_utility',
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'webrtc_video_coding',
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'<@(neteq_dependencies)',
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'<(rbe_components_path)/remote_bitrate_estimator_components.gyp:rbe_components_unittests',
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'<(DEPTH)/testing/gmock.gyp:gmock',
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'<(DEPTH)/testing/gtest.gyp:gtest',
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'<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
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@ -163,9 +164,6 @@
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'pacing/paced_sender_unittest.cc',
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'remote_bitrate_estimator/include/mock/mock_remote_bitrate_observer.h',
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'remote_bitrate_estimator/bitrate_estimator_unittest.cc',
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'remote_bitrate_estimator/remote_bitrate_estimator_single_stream_unittest.cc',
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'remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc',
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'remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h',
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'remote_bitrate_estimator/rtp_to_ntp_unittest.cc',
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'rtp_rtcp/source/mock/mock_rtp_payload_strategy.h',
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'rtp_rtcp/source/fec_test_helper.cc',
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@ -7,18 +7,16 @@
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# be found in the AUTHORS file in the root of the source tree.
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{
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'includes': [
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'../../build/common.gypi',
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],
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'targets': [
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{
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'target_name': 'remote_bitrate_estimator',
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'type': 'static_library',
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'dependencies': [
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# system_wrappers
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'<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
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],
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'include_dirs': [
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'include',
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'../rtp_rtcp/interface',
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'../interface',
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'<(rbe_components_path)/remote_bitrate_estimator_components.gyp:rbe_components',
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],
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'direct_dependent_settings': {
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'include_dirs': [
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@ -26,19 +24,11 @@
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],
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},
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'sources': [
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# interface
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'include/bwe_defines.h',
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'include/remote_bitrate_estimator.h',
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'include/rtp_to_ntp.h',
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# source
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'bitrate_estimator.cc',
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'bitrate_estimator.h',
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'overuse_detector.cc',
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'overuse_detector.h',
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'remote_bitrate_estimator_single_stream.cc',
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'remote_rate_control.cc',
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'remote_rate_control.h',
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'rtp_to_ntp.cc',
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], # source
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},
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@ -0,0 +1,45 @@
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# Copyright (c) 2013 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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{
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'includes': [
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'../../build/common.gypi',
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],
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'targets': [
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{
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'target_name': 'rbe_components',
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'type': 'static_library',
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'include_dirs': [
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'<(webrtc_root)/modules/remote_bitrate_estimator',
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],
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'sources': [
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'bitrate_estimator.cc',
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'bitrate_estimator.h',
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'overuse_detector.cc',
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'overuse_detector.h',
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'remote_bitrate_estimator_single_stream.cc',
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'remote_rate_control.cc',
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'remote_rate_control.h',
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],
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},
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{
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'target_name': 'rbe_components_unittests',
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'type': 'static_library',
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'dependencies': [
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'<(DEPTH)/testing/gmock.gyp:gmock',
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'<(DEPTH)/testing/gtest.gyp:gtest',
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'<(webrtc_root)/modules/modules.gyp:remote_bitrate_estimator',
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],
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'sources': [
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'<(rbe_components_path)/remote_bitrate_estimator_single_stream_unittest.cc',
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'<(rbe_components_path)/remote_bitrate_estimator_unittest_helper.cc',
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'<(rbe_components_path)/remote_bitrate_estimator_unittest_helper.h',
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],
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},
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],
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}
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@ -13,8 +13,8 @@
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'type': 'static_library',
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'dependencies': [
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'<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
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'<(webrtc_root)/modules/modules.gyp:remote_bitrate_estimator',
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'<(webrtc_root)/modules/modules.gyp:paced_sender',
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'<(webrtc_root)/modules/modules.gyp:remote_bitrate_estimator',
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],
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'include_dirs': [
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'../interface',
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