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

Issue 1250383003: Revert "Renamed the ACMDump to RtcEventLog and moved it to webrtc/video, since it is not specific t… (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: 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")
10 import("../../build/webrtc.gni") 11 import("../../build/webrtc.gni")
11 12
12 config("audio_coding_config") { 13 config("audio_coding_config") {
13 include_dirs = [ 14 include_dirs = [
14 "main/interface", 15 "main/interface",
15 "../interface", 16 "../interface",
16 ] 17 ]
17 } 18 }
18 19
19 source_set("audio_coding") { 20 source_set("audio_coding") {
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72 "../../common_audio", 73 "../../common_audio",
73 "../../system_wrappers", 74 "../../system_wrappers",
74 ] 75 ]
75 76
76 if (rtc_include_opus) { 77 if (rtc_include_opus) {
77 defines += [ "WEBRTC_CODEC_OPUS" ] 78 defines += [ "WEBRTC_CODEC_OPUS" ]
78 deps += [ ":webrtc_opus" ] 79 deps += [ ":webrtc_opus" ]
79 } 80 }
80 } 81 }
81 82
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
82 source_set("audio_decoder_interface") { 112 source_set("audio_decoder_interface") {
83 sources = [ 113 sources = [
84 "codecs/audio_decoder.cc", 114 "codecs/audio_decoder.cc",
85 "codecs/audio_decoder.h", 115 "codecs/audio_decoder.h",
86 ] 116 ]
87 configs += [ "../..:common_config" ] 117 configs += [ "../..:common_config" ]
88 public_configs = [ "../..:common_inherited_config" ] 118 public_configs = [ "../..:common_inherited_config" ]
89 deps = [ 119 deps = [
90 "../..:webrtc_common", 120 "../..:webrtc_common",
91 ] 121 ]
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767 "../../system_wrappers", 797 "../../system_wrappers",
768 ] 798 ]
769 799
770 defines = [] 800 defines = []
771 801
772 if (rtc_include_opus) { 802 if (rtc_include_opus) {
773 defines += [ "WEBRTC_CODEC_OPUS" ] 803 defines += [ "WEBRTC_CODEC_OPUS" ]
774 deps += [ ":webrtc_opus" ] 804 deps += [ ":webrtc_opus" ]
775 } 805 }
776 } 806 }
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