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Issue 2456363002: Added audio mixer and removed audio device module in AudioState::Config. (Closed)
Patch Set: Removed default initializer. Created 4 years, 1 month ago
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1 # Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 1 # Copyright (c) 2015 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 { 9 {
10 'includes': [ '../build/common.gypi', ], 10 'includes': [ '../build/common.gypi', ],
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92 }, 92 },
93 ] 93 ]
94 }], 94 }],
95 ], # conditions 95 ], # conditions
96 'targets': [ 96 'targets': [
97 { 97 {
98 'target_name': 'call_api', 98 'target_name': 'call_api',
99 'type': 'static_library', 99 'type': 'static_library',
100 'dependencies': [ 100 'dependencies': [
101 # TODO(kjellander): Add remaining dependencies when webrtc:4243 is done. 101 # TODO(kjellander): Add remaining dependencies when webrtc:4243 is done.
102 ':audio_mixer_api',
102 '<(webrtc_root)/base/base.gyp:rtc_base_approved', 103 '<(webrtc_root)/base/base.gyp:rtc_base_approved',
103 '<(webrtc_root)/common.gyp:webrtc_common', 104 '<(webrtc_root)/common.gyp:webrtc_common',
104 '<(webrtc_root)/modules/modules.gyp:audio_encoder_interface', 105 '<(webrtc_root)/modules/modules.gyp:audio_encoder_interface',
105 ], 106 ],
106 'sources': [ 107 'sources': [
107 'call/audio_receive_stream.h', 108 'call/audio_receive_stream.h',
108 'call/audio_send_stream.cc', 109 'call/audio_send_stream.cc',
109 'call/audio_send_stream.h', 110 'call/audio_send_stream.h',
110 'call/audio_sink.h', 111 'call/audio_sink.h',
111 'call/audio_state.h', 112 'call/audio_state.h',
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239 'type': 'static_library', 240 'type': 'static_library',
240 'dependencies': [ 241 'dependencies': [
241 '<(webrtc_root)/base/base.gyp:rtc_base_approved', 242 '<(webrtc_root)/base/base.gyp:rtc_base_approved',
242 ], 243 ],
243 'sources': [ 244 'sources': [
244 'audio/audio_mixer.h', 245 'audio/audio_mixer.h',
245 ], 246 ],
246 }, # target rtc_stats_api 247 }, # target rtc_stats_api
247 ], # targets 248 ], # targets
248 } 249 }
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