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Issue 2985263002: Reland of RTCVideoEncoder: Report H264 profile information to WebRTC (Closed)
Patch Set: Add default argument to SetDefaultVideoCodec Created 3 years, 4 months ago
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1 # Copyright 2014 The Chromium Authors. All rights reserved. 1 # Copyright 2014 The Chromium Authors. All rights reserved.
2 # Use of this source code is governed by a BSD-style license that can be 2 # Use of this source code is governed by a BSD-style license that can be
3 # found in the LICENSE file. 3 # found in the LICENSE file.
4 4
5 import("//build/config/chromecast_build.gni") 5 import("//build/config/chromecast_build.gni")
6 import("//build/config/features.gni") 6 import("//build/config/features.gni")
7 import("//build/config/ui.gni") 7 import("//build/config/ui.gni")
8 import("//build/split_static_library.gni") 8 import("//build/split_static_library.gni")
9 import("//content/common/features.gni") 9 import("//content/common/features.gni")
10 import("//media/media_options.gni") 10 import("//media/media_options.gni")
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770 770
771 deps += [ 771 deps += [
772 "//crypto", 772 "//crypto",
773 "//third_party/libvpx", 773 "//third_party/libvpx",
774 "//third_party/opus", 774 "//third_party/opus",
775 "//third_party/webrtc/api:libjingle_peerconnection", 775 "//third_party/webrtc/api:libjingle_peerconnection",
776 "//third_party/webrtc/api:rtc_stats_api", 776 "//third_party/webrtc/api:rtc_stats_api",
777 "//third_party/webrtc/api:video_frame_api", 777 "//third_party/webrtc/api:video_frame_api",
778 "//third_party/webrtc/api/audio_codecs:builtin_audio_decoder_factory", 778 "//third_party/webrtc/api/audio_codecs:builtin_audio_decoder_factory",
779 "//third_party/webrtc/api/audio_codecs:builtin_audio_encoder_factory", 779 "//third_party/webrtc/api/audio_codecs:builtin_audio_encoder_factory",
780 "//third_party/webrtc/common_video:common_video",
780 "//third_party/webrtc/media:rtc_media", 781 "//third_party/webrtc/media:rtc_media",
781 "//third_party/webrtc/media:rtc_media_base", 782 "//third_party/webrtc/media:rtc_media_base",
782 "//third_party/webrtc/modules/audio_device", 783 "//third_party/webrtc/modules/audio_device",
783 "//third_party/webrtc/modules/audio_processing", 784 "//third_party/webrtc/modules/audio_processing",
784 "//third_party/webrtc/modules/audio_processing/aec_dump", 785 "//third_party/webrtc/modules/audio_processing/aec_dump",
785 "//third_party/webrtc/modules/video_coding:webrtc_h264", 786 "//third_party/webrtc/modules/video_coding:webrtc_h264",
786 "//third_party/webrtc/p2p:libstunprober", 787 "//third_party/webrtc/p2p:libstunprober",
787 "//third_party/webrtc/p2p:rtc_p2p", 788 "//third_party/webrtc/p2p:rtc_p2p",
788 "//third_party/webrtc/pc:rtc_pc", 789 "//third_party/webrtc/pc:rtc_pc",
789 "//third_party/webrtc/rtc_base:rtc_base", 790 "//third_party/webrtc/rtc_base:rtc_base",
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997 # See comment at the top of //content/BUILD.gn for how this works. 998 # See comment at the top of //content/BUILD.gn for how this works.
998 group("for_content_tests") { 999 group("for_content_tests") {
999 visibility = [ "//content/test/*" ] 1000 visibility = [ "//content/test/*" ]
1000 1001
1001 if (!is_component_build) { 1002 if (!is_component_build) {
1002 public_deps = [ 1003 public_deps = [
1003 ":renderer", 1004 ":renderer",
1004 ] 1005 ]
1005 } 1006 }
1006 } 1007 }
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