BUG=1655 Review URL: https://webrtc-codereview.appspot.com/1326005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3857 4adac7df-926f-26a2-2b94-8c16560cd09d
89 lines
2.6 KiB
Python
89 lines
2.6 KiB
Python
# 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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{
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'targets': [
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{
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'target_name': 'webrtc_utility',
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'type': 'static_library',
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'dependencies': [
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'audio_coding_module',
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'<(webrtc_root)/common_audio/common_audio.gyp:resampler',
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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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'../interface',
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'../../interface',
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'../../media_file/interface',
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],
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'direct_dependent_settings': {
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'include_dirs': [
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'../interface',
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'../../interface',
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'../../audio_coding/main/interface',
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],
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},
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'sources': [
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'../interface/audio_frame_operations.h',
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'../interface/file_player.h',
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'../interface/file_recorder.h',
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'../interface/process_thread.h',
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'../interface/rtp_dump.h',
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'audio_frame_operations.cc',
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'coder.cc',
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'coder.h',
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'file_player_impl.cc',
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'file_player_impl.h',
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'file_recorder_impl.cc',
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'file_recorder_impl.h',
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'process_thread_impl.cc',
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'process_thread_impl.h',
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'rtp_dump_impl.cc',
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'rtp_dump_impl.h',
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],
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'conditions': [
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['enable_video==1', {
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# Adds support for video recording.
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'defines': [
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'WEBRTC_MODULE_UTILITY_VIDEO',
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],
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'dependencies': [
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'webrtc_video_coding',
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],
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'include_dirs': [
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'../../video_coding/main/interface',
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],
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'sources': [
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'frame_scaler.cc',
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'video_coder.cc',
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'video_frames_queue.cc',
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],
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}],
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],
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},
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], # targets
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'conditions': [
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['include_tests==1', {
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'targets': [
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{
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'target_name': 'webrtc_utility_unittests',
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'type': 'executable',
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'dependencies': [
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'webrtc_utility',
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'<(DEPTH)/testing/gtest.gyp:gtest',
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'<(webrtc_root)/test/test.gyp:test_support_main',
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],
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'sources': [
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'audio_frame_operations_unittest.cc',
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],
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}, # webrtc_utility_unittests
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], # targets
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}], # include_tests
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], # conditions
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}
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