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Moved gypi to avoid presubmit warning about '..' when touching the files.

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R=kjellander@webrtc.org,mflodman@webrtc.org
TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39299004

Cr-Commit-Position: refs/heads/master@{#8503}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8503 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
andresp@webrtc.org
2015-02-25 11:50:17 +00:00
parent 5c928ebd1d
commit 749c60217d
10 changed files with 284 additions and 294 deletions
Download Patch File Download Diff File Expand all files Collapse all files

36
webrtc/modules/audio_conference_mixer/source/audio_conference_mixer.gypi → webrtc/modules/audio_conference_mixer/audio_conference_mixer.gypi
View File

@@ -16,30 +16,20 @@
'webrtc_utility',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
],
'include_dirs': [
'../interface',
'../../interface',
],
'direct_dependent_settings': {
'include_dirs': [
'../interface',
'../../interface',
],
},
'sources': [
'../interface/audio_conference_mixer.h',
'../interface/audio_conference_mixer_defines.h',
'audio_frame_manipulator.cc',
'audio_frame_manipulator.h',
'level_indicator.cc',
'level_indicator.h',
'memory_pool.h',
'memory_pool_posix.h',
'memory_pool_win.h',
'audio_conference_mixer_impl.cc',
'audio_conference_mixer_impl.h',
'time_scheduler.cc',
'time_scheduler.h',
'interface/audio_conference_mixer.h',
'interface/audio_conference_mixer_defines.h',
'source/audio_frame_manipulator.cc',
'source/audio_frame_manipulator.h',
'source/level_indicator.cc',
'source/level_indicator.h',
'source/memory_pool.h',
'source/memory_pool_posix.h',
'source/memory_pool_win.h',
'source/audio_conference_mixer_impl.cc',
'source/audio_conference_mixer_impl.h',
'source/time_scheduler.cc',
'source/time_scheduler.h',
],
},
], # targets

12
webrtc/modules/modules.gyp
View File

@@ -10,7 +10,7 @@
'includes': [
'../build/common.gypi',
'audio_coding/audio_coding.gypi',
'audio_conference_mixer/source/audio_conference_mixer.gypi',
'audio_conference_mixer/audio_conference_mixer.gypi',
'audio_device/audio_device.gypi',
'audio_processing/audio_processing.gypi',
'bitrate_controller/bitrate_controller.gypi',
@@ -18,12 +18,12 @@
'media_file/media_file.gypi',
'pacing/pacing.gypi',
'remote_bitrate_estimator/remote_bitrate_estimator.gypi',
'rtp_rtcp/source/rtp_rtcp.gypi',
'utility/source/utility.gypi',
'rtp_rtcp/rtp_rtcp.gypi',
'utility/utility.gypi',
'video_coding/codecs/i420/main/source/i420.gypi',
'video_coding/main/source/video_coding.gypi',
'video_coding/video_coding.gypi',
'video_capture/video_capture.gypi',
'video_processing/main/source/video_processing.gypi',
'video_processing/video_processing.gypi',
'video_render/video_render.gypi',
],
'conditions': [
@@ -32,7 +32,7 @@
'audio_coding/audio_coding_tests.gypi',
'audio_processing/audio_processing_tests.gypi',
'rtp_rtcp/test/testFec/test_fec.gypi',
'video_coding/main/source/video_coding_test.gypi',
'video_coding/video_coding_test.gypi',
'video_coding/codecs/test/video_codecs_test_framework.gypi',
'video_coding/codecs/tools/video_codecs_tools.gypi',
], # includes

4
webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator.gypi
View File

@@ -60,7 +60,7 @@
'target_name': 'bwe_rtp_to_text',
'type': 'executable',
'includes': [
'../rtp_rtcp/source/rtp_rtcp.gypi',
'../rtp_rtcp/rtp_rtcp.gypi',
],
'dependencies': [
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
@@ -82,7 +82,7 @@
'target_name': 'bwe_rtp_play',
'type': 'executable',
'includes': [
'../rtp_rtcp/source/rtp_rtcp.gypi',
'../rtp_rtcp/rtp_rtcp.gypi',
],
'dependencies': [
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',

107
webrtc/modules/rtp_rtcp/rtp_rtcp.gypi Normal file
View File

@@ -0,0 +1,107 @@
# Copyright (c) 2011 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.
{
'targets': [
{
'target_name': 'rtp_rtcp',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
'<(webrtc_root)/modules/modules.gyp:paced_sender',
'<(webrtc_root)/modules/modules.gyp:remote_bitrate_estimator',
],
'sources': [
# Common
'interface/fec_receiver.h',
'interface/receive_statistics.h',
'interface/remote_ntp_time_estimator.h',
'interface/rtp_header_parser.h',
'interface/rtp_payload_registry.h',
'interface/rtp_receiver.h',
'interface/rtp_rtcp.h',
'interface/rtp_rtcp_defines.h',
'source/bitrate.cc',
'source/bitrate.h',
'source/byte_io.h',
'source/fec_receiver_impl.cc',
'source/fec_receiver_impl.h',
'source/receive_statistics_impl.cc',
'source/receive_statistics_impl.h',
'source/remote_ntp_time_estimator.cc',
'source/rtp_header_parser.cc',
'source/rtp_rtcp_config.h',
'source/rtp_rtcp_impl.cc',
'source/rtp_rtcp_impl.h',
'source/rtcp_packet.cc',
'source/rtcp_packet.h',
'source/rtcp_receiver.cc',
'source/rtcp_receiver.h',
'source/rtcp_receiver_help.cc',
'source/rtcp_receiver_help.h',
'source/rtcp_sender.cc',
'source/rtcp_sender.h',
'source/rtcp_utility.cc',
'source/rtcp_utility.h',
'source/rtp_header_extension.cc',
'source/rtp_header_extension.h',
'source/rtp_receiver_impl.cc',
'source/rtp_receiver_impl.h',
'source/rtp_sender.cc',
'source/rtp_sender.h',
'source/rtp_utility.cc',
'source/rtp_utility.h',
'source/ssrc_database.cc',
'source/ssrc_database.h',
'source/tmmbr_help.cc',
'source/tmmbr_help.h',
# Audio Files
'source/dtmf_queue.cc',
'source/dtmf_queue.h',
'source/rtp_receiver_audio.cc',
'source/rtp_receiver_audio.h',
'source/rtp_sender_audio.cc',
'source/rtp_sender_audio.h',
# Video Files
'source/fec_private_tables_random.h',
'source/fec_private_tables_bursty.h',
'source/forward_error_correction.cc',
'source/forward_error_correction.h',
'source/forward_error_correction_internal.cc',
'source/forward_error_correction_internal.h',
'source/producer_fec.cc',
'source/producer_fec.h',
'source/rtp_packet_history.cc',
'source/rtp_packet_history.h',
'source/rtp_payload_registry.cc',
'source/rtp_receiver_strategy.cc',
'source/rtp_receiver_strategy.h',
'source/rtp_receiver_video.cc',
'source/rtp_receiver_video.h',
'source/rtp_sender_video.cc',
'source/rtp_sender_video.h',
'source/video_codec_information.h',
'source/rtp_format.cc',
'source/rtp_format.h',
'source/rtp_format_h264.cc',
'source/rtp_format_h264.h',
'source/rtp_format_vp8.cc',
'source/rtp_format_vp8.h',
'source/rtp_format_video_generic.cc',
'source/rtp_format_video_generic.h',
'source/vp8_partition_aggregator.cc',
'source/vp8_partition_aggregator.h',
# Mocks
'mocks/mock_rtp_rtcp.h',
'source/mock/mock_rtp_payload_strategy.h',
], # source
# TODO(jschuh): Bug 1348: fix size_t to int truncations.
'msvs_disabled_warnings': [ 4267, ],
},
],
}

107
webrtc/modules/rtp_rtcp/source/rtp_rtcp.gypi
View File

@@ -1,107 +0,0 @@
# Copyright (c) 2011 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.
{
'targets': [
{
'target_name': 'rtp_rtcp',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
'<(webrtc_root)/modules/modules.gyp:paced_sender',
'<(webrtc_root)/modules/modules.gyp:remote_bitrate_estimator',
],
'sources': [
# Common
'../interface/fec_receiver.h',
'../interface/receive_statistics.h',
'../interface/remote_ntp_time_estimator.h',
'../interface/rtp_header_parser.h',
'../interface/rtp_payload_registry.h',
'../interface/rtp_receiver.h',
'../interface/rtp_rtcp.h',
'../interface/rtp_rtcp_defines.h',
'bitrate.cc',
'bitrate.h',
'byte_io.h',
'fec_receiver_impl.cc',
'fec_receiver_impl.h',
'receive_statistics_impl.cc',
'receive_statistics_impl.h',
'remote_ntp_time_estimator.cc',
'rtp_header_parser.cc',
'rtp_rtcp_config.h',
'rtp_rtcp_impl.cc',
'rtp_rtcp_impl.h',
'rtcp_packet.cc',
'rtcp_packet.h',
'rtcp_receiver.cc',
'rtcp_receiver.h',
'rtcp_receiver_help.cc',
'rtcp_receiver_help.h',
'rtcp_sender.cc',
'rtcp_sender.h',
'rtcp_utility.cc',
'rtcp_utility.h',
'rtp_header_extension.cc',
'rtp_header_extension.h',
'rtp_receiver_impl.cc',
'rtp_receiver_impl.h',
'rtp_sender.cc',
'rtp_sender.h',
'rtp_utility.cc',
'rtp_utility.h',
'ssrc_database.cc',
'ssrc_database.h',
'tmmbr_help.cc',
'tmmbr_help.h',
# Audio Files
'dtmf_queue.cc',
'dtmf_queue.h',
'rtp_receiver_audio.cc',
'rtp_receiver_audio.h',
'rtp_sender_audio.cc',
'rtp_sender_audio.h',
# Video Files
'fec_private_tables_random.h',
'fec_private_tables_bursty.h',
'forward_error_correction.cc',
'forward_error_correction.h',
'forward_error_correction_internal.cc',
'forward_error_correction_internal.h',
'producer_fec.cc',
'producer_fec.h',
'rtp_packet_history.cc',
'rtp_packet_history.h',
'rtp_payload_registry.cc',
'rtp_receiver_strategy.cc',
'rtp_receiver_strategy.h',
'rtp_receiver_video.cc',
'rtp_receiver_video.h',
'rtp_sender_video.cc',
'rtp_sender_video.h',
'video_codec_information.h',
'rtp_format.cc',
'rtp_format.h',
'rtp_format_h264.cc',
'rtp_format_h264.h',
'rtp_format_vp8.cc',
'rtp_format_vp8.h',
'rtp_format_video_generic.cc',
'rtp_format_video_generic.h',
'vp8_partition_aggregator.cc',
'vp8_partition_aggregator.h',
# Mocks
'../mocks/mock_rtp_rtcp.h',
'mock/mock_rtp_payload_strategy.h',
], # source
# TODO(jschuh): Bug 1348: fix size_t to int truncations.
'msvs_disabled_warnings': [ 4267, ],
},
],
}

42
webrtc/modules/utility/source/utility.gypi → webrtc/modules/utility/utility.gypi
View File

@@ -18,24 +18,24 @@
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
],
'sources': [
'../interface/audio_frame_operations.h',
'../interface/file_player.h',
'../interface/file_recorder.h',
'../interface/helpers_android.h',
'../interface/process_thread.h',
'../interface/rtp_dump.h',
'audio_frame_operations.cc',
'coder.cc',
'coder.h',
'file_player_impl.cc',
'file_player_impl.h',
'file_recorder_impl.cc',
'file_recorder_impl.h',
'helpers_android.cc',
'process_thread_impl.cc',
'process_thread_impl.h',
'rtp_dump_impl.cc',
'rtp_dump_impl.h',
'interface/audio_frame_operations.h',
'interface/file_player.h',
'interface/file_recorder.h',
'interface/helpers_android.h',
'interface/process_thread.h',
'interface/rtp_dump.h',
'source/audio_frame_operations.cc',
'source/coder.cc',
'source/coder.h',
'source/file_player_impl.cc',
'source/file_player_impl.h',
'source/file_recorder_impl.cc',
'source/file_recorder_impl.h',
'source/helpers_android.cc',
'source/process_thread_impl.cc',
'source/process_thread_impl.h',
'source/rtp_dump_impl.cc',
'source/rtp_dump_impl.h',
],
'conditions': [
['enable_video==1', {
@@ -43,9 +43,9 @@
'webrtc_video_coding',
],
'sources': [
'frame_scaler.cc',
'video_coder.cc',
'video_frames_queue.cc',
'source/frame_scaler.cc',
'source/video_coder.cc',
'source/video_frames_queue.cc',
],
}],
],

85
webrtc/modules/video_coding/main/source/video_coding.gypi
View File

@@ -1,85 +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.
{
'targets': [
{
'target_name': 'webrtc_video_coding',
'type': 'static_library',
'dependencies': [
'webrtc_i420',
'<(webrtc_root)/common_video/common_video.gyp:common_video',
'<(webrtc_root)/modules/video_coding/utility/video_coding_utility.gyp:video_coding_utility',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
'<(webrtc_vp8_dir)/vp8.gyp:webrtc_vp8',
'<(webrtc_vp9_dir)/vp9.gyp:webrtc_vp9',
],
'sources': [
# interfaces
'../interface/video_coding.h',
'../interface/video_coding_defines.h',
# headers
'codec_database.h',
'codec_timer.h',
'content_metrics_processing.h',
'decoding_state.h',
'encoded_frame.h',
'er_tables_xor.h',
'fec_tables_xor.h',
'frame_buffer.h',
'generic_decoder.h',
'generic_encoder.h',
'inter_frame_delay.h',
'internal_defines.h',
'jitter_buffer.h',
'jitter_buffer_common.h',
'jitter_estimator.h',
'media_opt_util.h',
'media_optimization.h',
'nack_fec_tables.h',
'packet.h',
'qm_select_data.h',
'qm_select.h',
'receiver.h',
'rtt_filter.h',
'session_info.h',
'timestamp_map.h',
'timing.h',
'video_coding_impl.h',
# sources
'codec_database.cc',
'codec_timer.cc',
'content_metrics_processing.cc',
'decoding_state.cc',
'encoded_frame.cc',
'frame_buffer.cc',
'generic_decoder.cc',
'generic_encoder.cc',
'inter_frame_delay.cc',
'jitter_buffer.cc',
'jitter_estimator.cc',
'media_opt_util.cc',
'media_optimization.cc',
'packet.cc',
'qm_select.cc',
'receiver.cc',
'rtt_filter.cc',
'session_info.cc',
'timestamp_map.cc',
'timing.cc',
'video_coding_impl.cc',
'video_sender.cc',
'video_receiver.cc',
], # source
# TODO(jschuh): Bug 1348: fix size_t to int truncations.
'msvs_disabled_warnings': [ 4267, ],
},
],
}

85
webrtc/modules/video_coding/video_coding.gypi Normal file
View File

@@ -0,0 +1,85 @@
# 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.
{
'targets': [
{
'target_name': 'webrtc_video_coding',
'type': 'static_library',
'dependencies': [
'webrtc_i420',
'<(webrtc_root)/common_video/common_video.gyp:common_video',
'<(webrtc_root)/modules/video_coding/utility/video_coding_utility.gyp:video_coding_utility',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
'<(webrtc_vp8_dir)/vp8.gyp:webrtc_vp8',
'<(webrtc_vp9_dir)/vp9.gyp:webrtc_vp9',
],
'sources': [
# interfaces
'main/interface/video_coding.h',
'main/interface/video_coding_defines.h',
# headers
'main/source/codec_database.h',
'main/source/codec_timer.h',
'main/source/content_metrics_processing.h',
'main/source/decoding_state.h',
'main/source/encoded_frame.h',
'main/source/er_tables_xor.h',
'main/source/fec_tables_xor.h',
'main/source/frame_buffer.h',
'main/source/generic_decoder.h',
'main/source/generic_encoder.h',
'main/source/inter_frame_delay.h',
'main/source/internal_defines.h',
'main/source/jitter_buffer.h',
'main/source/jitter_buffer_common.h',
'main/source/jitter_estimator.h',
'main/source/media_opt_util.h',
'main/source/media_optimization.h',
'main/source/nack_fec_tables.h',
'main/source/packet.h',
'main/source/qm_select_data.h',
'main/source/qm_select.h',
'main/source/receiver.h',
'main/source/rtt_filter.h',
'main/source/session_info.h',
'main/source/timestamp_map.h',
'main/source/timing.h',
'main/source/video_coding_impl.h',
# sources
'main/source/codec_database.cc',
'main/source/codec_timer.cc',
'main/source/content_metrics_processing.cc',
'main/source/decoding_state.cc',
'main/source/encoded_frame.cc',
'main/source/frame_buffer.cc',
'main/source/generic_decoder.cc',
'main/source/generic_encoder.cc',
'main/source/inter_frame_delay.cc',
'main/source/jitter_buffer.cc',
'main/source/jitter_estimator.cc',
'main/source/media_opt_util.cc',
'main/source/media_optimization.cc',
'main/source/packet.cc',
'main/source/qm_select.cc',
'main/source/receiver.cc',
'main/source/rtt_filter.cc',
'main/source/session_info.cc',
'main/source/timestamp_map.cc',
'main/source/timing.cc',
'main/source/video_coding_impl.cc',
'main/source/video_sender.cc',
'main/source/video_receiver.cc',
], # source
# TODO(jschuh): Bug 1348: fix size_t to int truncations.
'msvs_disabled_warnings': [ 4267, ],
},
],
}

56
webrtc/modules/video_coding/main/source/video_coding_test.gypi → webrtc/modules/video_coding/video_coding_test.gypi
View File

@@ -25,36 +25,36 @@
],
'sources': [
# headers
'../test/codec_database_test.h',
'../test/generic_codec_test.h',
'../test/media_opt_test.h',
'../test/mt_test_common.h',
'../test/normal_test.h',
'../test/quality_modes_test.h',
'../test/receiver_tests.h',
'../test/release_test.h',
'../test/rtp_player.h',
'../test/test_callbacks.h',
'../test/test_util.h',
'../test/vcm_payload_sink_factory.h',
'../test/video_source.h',
'main/test/codec_database_test.h',
'main/test/generic_codec_test.h',
'main/test/media_opt_test.h',
'main/test/mt_test_common.h',
'main/test/normal_test.h',
'main/test/quality_modes_test.h',
'main/test/receiver_tests.h',
'main/test/release_test.h',
'main/test/rtp_player.h',
'main/test/test_callbacks.h',
'main/test/test_util.h',
'main/test/vcm_payload_sink_factory.h',
'main/test/video_source.h',
# sources
'../test/codec_database_test.cc',
'../test/generic_codec_test.cc',
'../test/media_opt_test.cc',
'../test/mt_rx_tx_test.cc',
'../test/mt_test_common.cc',
'../test/normal_test.cc',
'../test/quality_modes_test.cc',
'../test/rtp_player.cc',
'../test/test_callbacks.cc',
'../test/test_util.cc',
'../test/tester_main.cc',
'../test/vcm_payload_sink_factory.cc',
'../test/video_rtp_play.cc',
'../test/video_rtp_play_mt.cc',
'../test/video_source.cc',
'main/test/codec_database_test.cc',
'main/test/generic_codec_test.cc',
'main/test/media_opt_test.cc',
'main/test/mt_rx_tx_test.cc',
'main/test/mt_test_common.cc',
'main/test/normal_test.cc',
'main/test/quality_modes_test.cc',
'main/test/rtp_player.cc',
'main/test/test_callbacks.cc',
'main/test/test_util.cc',
'main/test/tester_main.cc',
'main/test/vcm_payload_sink_factory.cc',
'main/test/video_rtp_play.cc',
'main/test/video_rtp_play_mt.cc',
'main/test/video_source.cc',
], # sources
},
],

44
webrtc/modules/video_processing/main/source/video_processing.gypi → webrtc/modules/video_processing/video_processing.gypi
View File

@@ -18,27 +18,27 @@
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
],
'sources': [
'../interface/video_processing.h',
'../interface/video_processing_defines.h',
'brighten.cc',
'brighten.h',
'brightness_detection.cc',
'brightness_detection.h',
'color_enhancement.cc',
'color_enhancement.h',
'color_enhancement_private.h',
'content_analysis.cc',
'content_analysis.h',
'deflickering.cc',
'deflickering.h',
'frame_preprocessor.cc',
'frame_preprocessor.h',
'spatial_resampler.cc',
'spatial_resampler.h',
'video_decimator.cc',
'video_decimator.h',
'video_processing_impl.cc',
'video_processing_impl.h',
'main/interface/video_processing.h',
'main/interface/video_processing_defines.h',
'main/source/brighten.cc',
'main/source/brighten.h',
'main/source/brightness_detection.cc',
'main/source/brightness_detection.h',
'main/source/color_enhancement.cc',
'main/source/color_enhancement.h',
'main/source/color_enhancement_private.h',
'main/source/content_analysis.cc',
'main/source/content_analysis.h',
'main/source/deflickering.cc',
'main/source/deflickering.h',
'main/source/frame_preprocessor.cc',
'main/source/frame_preprocessor.h',
'main/source/spatial_resampler.cc',
'main/source/spatial_resampler.h',
'main/source/video_decimator.cc',
'main/source/video_decimator.h',
'main/source/video_processing_impl.cc',
'main/source/video_processing_impl.h',
],
'conditions': [
['target_arch=="ia32" or target_arch=="x64"', {
@@ -54,7 +54,7 @@
'target_name': 'video_processing_sse2',
'type': 'static_library',
'sources': [
'content_analysis_sse2.cc',
'main/source/content_analysis_sse2.cc',
],
'conditions': [
['os_posix==1 and OS!="mac"', {