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Issue 1250513006: Remove webrtc::Config from ViEChannelGroup. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: remove experiment.h from build files Created 5 years, 5 months ago
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1 # Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 1 # Copyright (c) 2012 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 'conditions': [ 9 'conditions': [
10 ['include_tests==1', { 10 ['include_tests==1', {
(...skipping 50 matching lines...) Expand 10 before | Expand all | Expand 10 after
61 ], 61 ],
62 }, 62 },
63 { 63 {
64 'target_name': 'webrtc', 64 'target_name': 'webrtc',
65 'type': 'static_library', 65 'type': 'static_library',
66 'sources': [ 66 'sources': [
67 'audio_receive_stream.h', 67 'audio_receive_stream.h',
68 'audio_send_stream.h', 68 'audio_send_stream.h',
69 'call.h', 69 'call.h',
70 'config.h', 70 'config.h',
71 'experiments.h',
72 'frame_callback.h', 71 'frame_callback.h',
73 'stream.h', 72 'stream.h',
74 'transport.h', 73 'transport.h',
75 'video_receive_stream.h', 74 'video_receive_stream.h',
76 'video_renderer.h', 75 'video_renderer.h',
77 'video_send_stream.h', 76 'video_send_stream.h',
78 77
79 '<@(webrtc_video_sources)', 78 '<@(webrtc_video_sources)',
80 ], 79 ],
81 'dependencies': [ 80 'dependencies': [
82 'common.gyp:*', 81 'common.gyp:*',
83 '<@(webrtc_video_dependencies)', 82 '<@(webrtc_video_dependencies)',
84 ], 83 ],
85 'conditions': [ 84 'conditions': [
86 # TODO(andresp): Chromium libpeerconnection should link directly with 85 # TODO(andresp): Chromium libpeerconnection should link directly with
87 # this and no if conditions should be needed on webrtc build files. 86 # this and no if conditions should be needed on webrtc build files.
88 ['build_with_chromium==1', { 87 ['build_with_chromium==1', {
89 'dependencies': [ 88 'dependencies': [
90 '<(webrtc_root)/modules/modules.gyp:video_capture', 89 '<(webrtc_root)/modules/modules.gyp:video_capture',
91 '<(webrtc_root)/modules/modules.gyp:video_render', 90 '<(webrtc_root)/modules/modules.gyp:video_render',
92 ], 91 ],
93 }], 92 }],
94 ], 93 ],
95 }, 94 },
96 ], 95 ],
97 } 96 }
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