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Issue 2446963003: Clean up logging in AudioSendStream::SetupSendCodec(). (Closed)
Patch Set: fix build breakage? 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', ],
(...skipping 87 matching lines...) Expand 10 before | Expand all | Expand 10 after
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 '<(webrtc_root)/base/base.gyp:rtc_base_approved', 102 '<(webrtc_root)/base/base.gyp:rtc_base_approved',
103 '<(webrtc_root)/common.gyp:webrtc_common', 103 '<(webrtc_root)/common.gyp:webrtc_common',
104 '<(webrtc_root)/modules/modules.gyp:audio_encoder_interface', 104 '<(webrtc_root)/modules/modules.gyp:audio_encoder_interface',
105 ], 105 ],
106 'sources': [ 106 'sources': [
107 'call/audio_receive_stream.h', 107 'call/audio_receive_stream.h',
108 'call/audio_send_stream.cc',
108 'call/audio_send_stream.h', 109 'call/audio_send_stream.h',
109 'call/audio_sink.h', 110 'call/audio_sink.h',
110 'call/audio_state.h', 111 'call/audio_state.h',
111 'call/flexfec_receive_stream.h' 112 'call/flexfec_receive_stream.h'
112 ], 113 ],
113 }, 114 },
114 { 115 {
115 'target_name': 'libjingle_peerconnection', 116 'target_name': 'libjingle_peerconnection',
116 'type': 'static_library', 117 'type': 'static_library',
117 'dependencies': [ 118 'dependencies': [
(...skipping 120 matching lines...) Expand 10 before | Expand all | Expand 10 after
238 'type': 'static_library', 239 'type': 'static_library',
239 'dependencies': [ 240 'dependencies': [
240 '<(webrtc_root)/base/base.gyp:rtc_base_approved', 241 '<(webrtc_root)/base/base.gyp:rtc_base_approved',
241 ], 242 ],
242 'sources': [ 243 'sources': [
243 'audio/audio_mixer.h', 244 'audio/audio_mixer.h',
244 ], 245 ],
245 }, # target rtc_stats_api 246 }, # target rtc_stats_api
246 ], # targets 247 ], # targets
247 } 248 }
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