diff --git a/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer.gypi b/webrtc/modules/audio_conference_mixer/audio_conference_mixer.gypi similarity index 50% rename from webrtc/modules/audio_conference_mixer/source/audio_conference_mixer.gypi rename to webrtc/modules/audio_conference_mixer/audio_conference_mixer.gypi index 83d9a80f1..51ee6891d 100644 --- a/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer.gypi +++ b/webrtc/modules/audio_conference_mixer/audio_conference_mixer.gypi @@ -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 diff --git a/webrtc/modules/modules.gyp b/webrtc/modules/modules.gyp index 8bb3745dc..84789462d 100644 --- a/webrtc/modules/modules.gyp +++ b/webrtc/modules/modules.gyp @@ -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 diff --git a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator.gypi b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator.gypi index c3853f635..fd198989e 100644 --- a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator.gypi +++ b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator.gypi @@ -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', diff --git a/webrtc/modules/rtp_rtcp/rtp_rtcp.gypi b/webrtc/modules/rtp_rtcp/rtp_rtcp.gypi new file mode 100644 index 000000000..7a144e4ef --- /dev/null +++ b/webrtc/modules/rtp_rtcp/rtp_rtcp.gypi @@ -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, ], + }, + ], +} diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp.gypi b/webrtc/modules/rtp_rtcp/source/rtp_rtcp.gypi deleted file mode 100644 index 65f33e83d..000000000 --- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp.gypi +++ /dev/null @@ -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, ], - }, - ], -} diff --git a/webrtc/modules/utility/source/utility.gypi b/webrtc/modules/utility/utility.gypi similarity index 54% rename from webrtc/modules/utility/source/utility.gypi rename to webrtc/modules/utility/utility.gypi index c6e50463c..c39a18f90 100644 --- a/webrtc/modules/utility/source/utility.gypi +++ b/webrtc/modules/utility/utility.gypi @@ -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', ], }], ], diff --git a/webrtc/modules/video_coding/main/source/video_coding.gypi b/webrtc/modules/video_coding/main/source/video_coding.gypi deleted file mode 100644 index 4793a138e..000000000 --- a/webrtc/modules/video_coding/main/source/video_coding.gypi +++ /dev/null @@ -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, ], - }, - ], -} diff --git a/webrtc/modules/video_coding/video_coding.gypi b/webrtc/modules/video_coding/video_coding.gypi new file mode 100644 index 000000000..484b82a54 --- /dev/null +++ b/webrtc/modules/video_coding/video_coding.gypi @@ -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, ], + }, + ], +} diff --git a/webrtc/modules/video_coding/main/source/video_coding_test.gypi b/webrtc/modules/video_coding/video_coding_test.gypi similarity index 51% rename from webrtc/modules/video_coding/main/source/video_coding_test.gypi rename to webrtc/modules/video_coding/video_coding_test.gypi index 81fd4e3c8..c990c3a7a 100644 --- a/webrtc/modules/video_coding/main/source/video_coding_test.gypi +++ b/webrtc/modules/video_coding/video_coding_test.gypi @@ -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 }, ], diff --git a/webrtc/modules/video_processing/main/source/video_processing.gypi b/webrtc/modules/video_processing/video_processing.gypi similarity index 60% rename from webrtc/modules/video_processing/main/source/video_processing.gypi rename to webrtc/modules/video_processing/video_processing.gypi index 0126519bc..43c49b111 100644 --- a/webrtc/modules/video_processing/main/source/video_processing.gypi +++ b/webrtc/modules/video_processing/video_processing.gypi @@ -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"', {