
What used to be the libpeerconnection library is now compiled statically into the Chromium binary, so clean up references it. BUG=chromium:482123 TBR=tommi@webrtc.org Review URL: https://codereview.webrtc.org/1399513002 . Cr-Commit-Position: refs/heads/master@{#10216}
155 lines
4.3 KiB
Python
155 lines
4.3 KiB
Python
# Copyright (c) 2012 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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'conditions': [
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['include_tests==1', {
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'includes': [
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'libjingle/xmllite/xmllite_tests.gypi',
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'libjingle/xmpp/xmpp_tests.gypi',
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'p2p/p2p_tests.gypi',
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'sound/sound_tests.gypi',
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'webrtc_tests.gypi',
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],
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}],
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['enable_protobuf==1', {
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'targets': [
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{
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# This target should only be built if enable_protobuf is defined
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'target_name': 'rtc_event_log_proto',
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'type': 'static_library',
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'sources': ['call/rtc_event_log.proto',],
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'variables': {
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'proto_in_dir': 'call',
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'proto_out_dir': 'webrtc/call',
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},
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'includes': ['build/protoc.gypi'],
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},
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],
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}],
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['include_tests==1 and enable_protobuf==1', {
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'targets': [
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{
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'target_name': 'rtc_event_log2rtp_dump',
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'type': 'executable',
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'sources': ['call/rtc_event_log2rtp_dump.cc',],
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'dependencies': [
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'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
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'rtc_event_log',
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'rtc_event_log_proto',
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'test/test.gyp:rtp_test_utils'
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],
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},
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],
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}],
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],
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'includes': [
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'build/common.gypi',
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'audio/webrtc_audio.gypi',
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'call/webrtc_call.gypi',
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'video/webrtc_video.gypi',
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],
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'variables': {
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'webrtc_all_dependencies': [
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'base/base.gyp:*',
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'sound/sound.gyp:*',
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'common.gyp:*',
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'common_audio/common_audio.gyp:*',
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'common_video/common_video.gyp:*',
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'modules/modules.gyp:*',
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'p2p/p2p.gyp:*',
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'system_wrappers/system_wrappers.gyp:*',
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'tools/tools.gyp:*',
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'voice_engine/voice_engine.gyp:*',
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'<(webrtc_vp8_dir)/vp8.gyp:*',
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'<(webrtc_vp9_dir)/vp9.gyp:*',
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],
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},
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'targets': [
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{
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'target_name': 'webrtc_all',
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'type': 'none',
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'dependencies': [
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'<@(webrtc_all_dependencies)',
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'webrtc',
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],
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'conditions': [
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['include_tests==1', {
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'dependencies': [
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'common_video/common_video_unittests.gyp:*',
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'rtc_unittests',
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'system_wrappers/system_wrappers_tests.gyp:*',
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'test/metrics.gyp:*',
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'test/test.gyp:*',
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'test/webrtc_test_common.gyp:*',
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'video_engine/video_engine_core_unittests.gyp:*',
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'webrtc_tests',
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],
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}],
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],
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},
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{
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'target_name': 'webrtc',
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'type': 'static_library',
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'sources': [
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'audio_receive_stream.h',
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'audio_send_stream.h',
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'call.h',
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'config.h',
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'frame_callback.h',
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'stream.h',
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'transport.h',
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'video_receive_stream.h',
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'video_renderer.h',
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'video_send_stream.h',
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'<@(webrtc_audio_sources)',
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'<@(webrtc_call_sources)',
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'<@(webrtc_video_sources)',
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],
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'dependencies': [
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'common.gyp:*',
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'<@(webrtc_audio_dependencies)',
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'<@(webrtc_call_dependencies)',
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'<@(webrtc_video_dependencies)',
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'rtc_event_log',
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],
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'conditions': [
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# TODO(andresp): Chromium should link directly with this and no if
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# conditions should be needed on webrtc build files.
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['build_with_chromium==1', {
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'dependencies': [
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'<(webrtc_root)/modules/modules.gyp:video_capture',
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'<(webrtc_root)/modules/modules.gyp:video_render',
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],
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}],
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],
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},
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{
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'target_name': 'rtc_event_log',
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'type': 'static_library',
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'sources': [
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'call/rtc_event_log.cc',
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'call/rtc_event_log.h',
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],
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'conditions': [
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# If enable_protobuf is defined, we want to compile the protobuf
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# and add rtc_event_log.pb.h and rtc_event_log.pb.cc to the sources.
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['enable_protobuf==1', {
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'dependencies': [
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'rtc_event_log_proto',
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],
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'defines': [
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'ENABLE_RTC_EVENT_LOG',
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],
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}],
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],
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},
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],
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}
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