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Issue 2260803002: Generalize FEC header formatting. (pt. 4) (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Lint fix. Created 4 years, 3 months ago
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1 # Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. 1 # Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
2 # 2 #
3 # Use of this source code is governed by a BSD-style license 3 # Use of this source code is governed by a BSD-style license
4 # that can be found in the LICENSE file in the root of the source 4 # that can be found in the LICENSE file in the root of the source
5 # tree. An additional intellectual property rights grant can be found 5 # tree. An additional intellectual property rights grant can be found
6 # in the file PATENTS. All contributing project authors may 6 # in the file PATENTS. All contributing project authors may
7 # be found in the AUTHORS file in the root of the source tree. 7 # be found in the AUTHORS file in the root of the source tree.
8 8
9 import("../build/webrtc.gni") 9 import("../build/webrtc.gni")
10 import("audio_coding/audio_coding.gni") 10 import("audio_coding/audio_coding.gni")
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438 "rtp_rtcp/source/rtp_format_vp8_test_helper.h", 438 "rtp_rtcp/source/rtp_format_vp8_test_helper.h",
439 "rtp_rtcp/source/rtp_format_vp8_unittest.cc", 439 "rtp_rtcp/source/rtp_format_vp8_unittest.cc",
440 "rtp_rtcp/source/rtp_format_vp9_unittest.cc", 440 "rtp_rtcp/source/rtp_format_vp9_unittest.cc",
441 "rtp_rtcp/source/rtp_header_extension_unittest.cc", 441 "rtp_rtcp/source/rtp_header_extension_unittest.cc",
442 "rtp_rtcp/source/rtp_packet_history_unittest.cc", 442 "rtp_rtcp/source/rtp_packet_history_unittest.cc",
443 "rtp_rtcp/source/rtp_packet_unittest.cc", 443 "rtp_rtcp/source/rtp_packet_unittest.cc",
444 "rtp_rtcp/source/rtp_payload_registry_unittest.cc", 444 "rtp_rtcp/source/rtp_payload_registry_unittest.cc",
445 "rtp_rtcp/source/rtp_rtcp_impl_unittest.cc", 445 "rtp_rtcp/source/rtp_rtcp_impl_unittest.cc",
446 "rtp_rtcp/source/rtp_sender_unittest.cc", 446 "rtp_rtcp/source/rtp_sender_unittest.cc",
447 "rtp_rtcp/source/time_util_unittest.cc", 447 "rtp_rtcp/source/time_util_unittest.cc",
448 "rtp_rtcp/source/ulpfec_header_reader_writer_unittest.cc",
448 "rtp_rtcp/source/vp8_partition_aggregator_unittest.cc", 449 "rtp_rtcp/source/vp8_partition_aggregator_unittest.cc",
449 "rtp_rtcp/test/testAPI/test_api.cc", 450 "rtp_rtcp/test/testAPI/test_api.cc",
450 "rtp_rtcp/test/testAPI/test_api.h", 451 "rtp_rtcp/test/testAPI/test_api.h",
451 "rtp_rtcp/test/testAPI/test_api_audio.cc", 452 "rtp_rtcp/test/testAPI/test_api_audio.cc",
452 "rtp_rtcp/test/testAPI/test_api_rtcp.cc", 453 "rtp_rtcp/test/testAPI/test_api_rtcp.cc",
453 "rtp_rtcp/test/testAPI/test_api_video.cc", 454 "rtp_rtcp/test/testAPI/test_api_video.cc",
454 "utility/source/audio_frame_operations_unittest.cc", 455 "utility/source/audio_frame_operations_unittest.cc",
455 "utility/source/file_player_unittests.cc", 456 "utility/source/file_player_unittests.cc",
456 "utility/source/process_thread_impl_unittest.cc", 457 "utility/source/process_thread_impl_unittest.cc",
457 "video_coding/codecs/test/packet_manipulator_unittest.cc", 458 "video_coding/codecs/test/packet_manipulator_unittest.cc",
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687 sources += [ 688 sources += [
688 "audio_device/ios/audio_device_unittest_ios.cc", 689 "audio_device/ios/audio_device_unittest_ios.cc",
689 "audio_device/ios/objc/RTCAudioSessionTest.mm", 690 "audio_device/ios/objc/RTCAudioSessionTest.mm",
690 "video_coding/codecs/h264/h264_video_toolbox_nalu_unittest.cc", 691 "video_coding/codecs/h264/h264_video_toolbox_nalu_unittest.cc",
691 ] 692 ]
692 693
693 ldflags = [ "-ObjC" ] 694 ldflags = [ "-ObjC" ]
694 } 695 }
695 } 696 }
696 } 697 }
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