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

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

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.h',
-        '../interface/audio_conference_mixer_defines.h',
+        'interface/audio_conference_mixer_defines.h',
-        'audio_frame_manipulator.cc',
+        'source/audio_frame_manipulator.cc',
-        'audio_frame_manipulator.h',
+        'source/audio_frame_manipulator.h',
-        'level_indicator.cc',
+        'source/level_indicator.cc',
-        'level_indicator.h',
+        'source/level_indicator.h',
-        'memory_pool.h',
+        'source/memory_pool.h',
-        'memory_pool_posix.h',
+        'source/memory_pool_posix.h',
-        'memory_pool_win.h',
+        'source/memory_pool_win.h',
-        'audio_conference_mixer_impl.cc',
+        'source/audio_conference_mixer_impl.cc',
-        'audio_conference_mixer_impl.h',
+        'source/audio_conference_mixer_impl.h',
-        'time_scheduler.cc',
+        'source/time_scheduler.cc',
-        'time_scheduler.h',
+        'source/time_scheduler.h',
       ],
     },
   ], # targets

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',
+    'rtp_rtcp/rtp_rtcp.gypi',
-    'utility/source/utility.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

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',

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, ],
+    },
+  ],
+}

webrtc/modules/rtp_rtcp/source/rtp_rtcp.gypi

@@ -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, ],
-    },
-  ],
-}

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

webrtc/modules/video_coding/main/source/video_coding.gypi

@@ -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, ],
-    },
-  ],
-}

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, ],
+    },
+  ],
+}

webrtc/modules/video_coding/video_coding_test.gypi

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

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',
+        'main/interface/video_processing.h',
-        '../interface/video_processing_defines.h',
+        'main/interface/video_processing_defines.h',
-        'brighten.cc',
+        'main/source/brighten.cc',
-        'brighten.h',
+        'main/source/brighten.h',
-        'brightness_detection.cc',
+        'main/source/brightness_detection.cc',
-        'brightness_detection.h',
+        'main/source/brightness_detection.h',
-        'color_enhancement.cc',
+        'main/source/color_enhancement.cc',
-        'color_enhancement.h',
+        'main/source/color_enhancement.h',
-        'color_enhancement_private.h',
+        'main/source/color_enhancement_private.h',
-        'content_analysis.cc',
+        'main/source/content_analysis.cc',
-        'content_analysis.h',
+        'main/source/content_analysis.h',
-        'deflickering.cc',
+        'main/source/deflickering.cc',
-        'deflickering.h',
+        'main/source/deflickering.h',
-        'frame_preprocessor.cc',
+        'main/source/frame_preprocessor.cc',
-        'frame_preprocessor.h',
+        'main/source/frame_preprocessor.h',
-        'spatial_resampler.cc',
+        'main/source/spatial_resampler.cc',
-        'spatial_resampler.h',
+        'main/source/spatial_resampler.h',
-        'video_decimator.cc',
+        'main/source/video_decimator.cc',
-        'video_decimator.h',
+        'main/source/video_decimator.h',
-        'video_processing_impl.cc',
+        'main/source/video_processing_impl.cc',
-        'video_processing_impl.h',
+        '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"', {