Index: webrtc/webrtc.gyp |
diff --git a/webrtc/webrtc.gyp b/webrtc/webrtc.gyp |
deleted file mode 100644 |
index 1447a99183aeee054c53b8019a09f28499b0fde7..0000000000000000000000000000000000000000 |
--- a/webrtc/webrtc.gyp |
+++ /dev/null |
@@ -1,130 +0,0 @@ |
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
-# |
-# Use of this source code is governed by a BSD-style license |
-# that can be found in the LICENSE file in the root of the source |
-# tree. An additional intellectual property rights grant can be found |
-# in the file PATENTS. All contributing project authors may |
-# be found in the AUTHORS file in the root of the source tree. |
-{ |
- 'includes': [ |
- 'build/common.gypi', |
- 'audio/webrtc_audio.gypi', |
- 'call/webrtc_call.gypi', |
- 'video/webrtc_video.gypi', |
- ], |
- 'targets': [ |
- { |
- 'target_name': 'webrtc', |
- 'type': 'static_library', |
- 'sources': [ |
- 'call.h', |
- 'config.h', |
- 'transport.h', |
- 'video_receive_stream.h', |
- 'video_send_stream.h', |
- |
- '<@(webrtc_audio_sources)', |
- '<@(webrtc_call_sources)', |
- '<@(webrtc_video_sources)', |
- ], |
- 'dependencies': [ |
- 'common.gyp:*', |
- '<@(webrtc_audio_dependencies)', |
- '<@(webrtc_call_dependencies)', |
- '<@(webrtc_video_dependencies)', |
- 'rtc_event_log_impl', |
- ], |
- 'conditions': [ |
- # TODO(andresp): Chromium should link directly with this and no if |
- # conditions should be needed on webrtc build files. |
- ['build_with_chromium==1', { |
- 'dependencies': [ |
- '<(webrtc_root)/modules/modules.gyp:video_capture', |
- ], |
- }], |
- ], |
- }, |
- { |
- 'target_name': 'rtc_event_log_api', |
- 'type': 'static_library', |
- 'sources': [ |
- 'logging/rtc_event_log/rtc_event_log.h', |
- ], |
- }, |
- { |
- 'target_name': 'rtc_event_log_impl', |
- 'type': 'static_library', |
- 'sources': [ |
- 'logging/rtc_event_log/ringbuffer.h', |
- 'logging/rtc_event_log/rtc_event_log.cc', |
- 'logging/rtc_event_log/rtc_event_log_helper_thread.cc', |
- 'logging/rtc_event_log/rtc_event_log_helper_thread.h', |
- ], |
- 'conditions': [ |
- # If enable_protobuf is defined, we want to compile the protobuf |
- # and add rtc_event_log.pb.h and rtc_event_log.pb.cc to the sources. |
- ['enable_protobuf==1', { |
- 'dependencies': [ |
- 'rtc_event_log_api', |
- 'rtc_event_log_proto', |
- '<(webrtc_root)/api/api.gyp:call_api', |
- ], |
- 'defines': [ |
- 'ENABLE_RTC_EVENT_LOG', |
- ], |
- }], |
- ], |
- }, |
- ], # targets |
- 'conditions': [ |
- ['include_tests==1', { |
- 'includes': [ |
- 'webrtc_tests.gypi', |
- ], |
- }], |
- ['enable_protobuf==1', { |
- 'targets': [ |
- { |
- # This target should only be built if enable_protobuf is defined |
- 'target_name': 'rtc_event_log_proto', |
- 'type': 'static_library', |
- 'sources': ['logging/rtc_event_log/rtc_event_log.proto',], |
- 'variables': { |
- 'proto_in_dir': 'logging/rtc_event_log', |
- 'proto_out_dir': 'webrtc/logging/rtc_event_log', |
- }, |
- 'includes': ['build/protoc.gypi'], |
- }, |
- { |
- 'target_name': 'rtc_event_log_parser', |
- 'type': 'static_library', |
- 'sources': [ |
- 'logging/rtc_event_log/rtc_event_log_parser.cc', |
- 'logging/rtc_event_log/rtc_event_log_parser.h', |
- ], |
- 'dependencies': [ |
- 'rtc_event_log_proto', |
- ], |
- 'export_dependent_settings': [ |
- 'rtc_event_log_proto', |
- ], |
- }, |
- ], |
- }], |
- ['include_tests==1 and enable_protobuf==1', { |
- 'targets': [ |
- { |
- 'target_name': 'rtc_event_log2rtp_dump', |
- 'type': 'executable', |
- 'sources': ['logging/rtc_event_log2rtp_dump.cc',], |
- 'dependencies': [ |
- '<(DEPTH)/third_party/gflags/gflags.gyp:gflags', |
- 'rtc_event_log_parser', |
- 'rtc_event_log_proto', |
- 'test/test.gyp:rtp_test_utils' |
- ], |
- }, |
- ], |
- }], |
- ], # conditions |
-} |