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Side by Side Diff: webrtc/modules/audio_coding/BUILD.gn

Issue 1230973005: Adds logging of configuration information for VideoReceiveStream (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Addressed comments from stefan. Created 5 years, 4 months ago
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1 # Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 1 # Copyright (c) 2014 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/config/arm.gni") 9 import("//build/config/arm.gni")
10 import("//third_party/protobuf/proto_library.gni")
11 import("../../build/webrtc.gni") 10 import("../../build/webrtc.gni")
12 11
13 config("audio_coding_config") { 12 config("audio_coding_config") {
14 include_dirs = [ 13 include_dirs = [
15 "main/interface", 14 "main/interface",
16 "../interface", 15 "../interface",
17 ] 16 ]
18 } 17 }
19 18
20 source_set("audio_coding") { 19 source_set("audio_coding") {
(...skipping 52 matching lines...) Expand 10 before | Expand all | Expand 10 after
73 "../../common_audio", 72 "../../common_audio",
74 "../../system_wrappers", 73 "../../system_wrappers",
75 ] 74 ]
76 75
77 if (rtc_include_opus) { 76 if (rtc_include_opus) {
78 defines += [ "WEBRTC_CODEC_OPUS" ] 77 defines += [ "WEBRTC_CODEC_OPUS" ]
79 deps += [ ":webrtc_opus" ] 78 deps += [ ":webrtc_opus" ]
80 } 79 }
81 } 80 }
82 81
83 proto_library("acm_dump_proto") {
84 sources = [
85 "main/acm2/dump.proto",
86 ]
87 proto_out_dir = "webrtc/audio_coding"
88 }
89
90 source_set("acm_dump") {
91 sources = [
92 "main/acm2/acm_dump.cc",
93 "main/acm2/acm_dump.h",
94 ]
95
96 defines = []
97
98 configs += [ "../..:common_config" ]
99
100 public_configs = [ "../..:common_inherited_config" ]
101
102 deps = [
103 ":acm_dump_proto",
104 "../..:webrtc_common",
105 ]
106
107 if (rtc_enable_protobuf) {
108 defines += [ "RTC_AUDIOCODING_DEBUG_DUMP" ]
109 }
110 }
111
112 source_set("audio_decoder_interface") { 82 source_set("audio_decoder_interface") {
113 sources = [ 83 sources = [
114 "codecs/audio_decoder.cc", 84 "codecs/audio_decoder.cc",
115 "codecs/audio_decoder.h", 85 "codecs/audio_decoder.h",
116 ] 86 ]
117 configs += [ "../..:common_config" ] 87 configs += [ "../..:common_config" ]
118 public_configs = [ "../..:common_inherited_config" ] 88 public_configs = [ "../..:common_inherited_config" ]
119 deps = [ 89 deps = [
120 "../..:webrtc_common", 90 "../..:webrtc_common",
121 ] 91 ]
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802 "../../system_wrappers", 772 "../../system_wrappers",
803 ] 773 ]
804 774
805 defines = [] 775 defines = []
806 776
807 if (rtc_include_opus) { 777 if (rtc_include_opus) {
808 defines += [ "WEBRTC_CODEC_OPUS" ] 778 defines += [ "WEBRTC_CODEC_OPUS" ]
809 deps += [ ":webrtc_opus" ] 779 deps += [ ":webrtc_opus" ]
810 } 780 }
811 } 781 }
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