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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") { |
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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 deps = [ | |
99 ":acm_dump_proto", | |
100 ] | |
101 | |
102 if (rtc_enable_protobuf) { | |
103 defines += [ "RTC_AUDIOCODING_DEBUG_DUMP" ] | |
104 } | |
105 } | |
106 | |
107 source_set("audio_decoder_interface") { | 82 source_set("audio_decoder_interface") { |
108 sources = [ | 83 sources = [ |
109 "codecs/audio_decoder.cc", | 84 "codecs/audio_decoder.cc", |
110 "codecs/audio_decoder.h", | 85 "codecs/audio_decoder.h", |
111 ] | 86 ] |
112 configs += [ "../..:common_config" ] | 87 configs += [ "../..:common_config" ] |
113 public_configs = [ "../..:common_inherited_config" ] | 88 public_configs = [ "../..:common_inherited_config" ] |
114 deps = [ | 89 deps = [ |
115 "../..:webrtc_common", | 90 "../..:webrtc_common", |
116 ] | 91 ] |
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808 "../../system_wrappers", | 783 "../../system_wrappers", |
809 ] | 784 ] |
810 | 785 |
811 defines = [] | 786 defines = [] |
812 | 787 |
813 if (rtc_include_opus) { | 788 if (rtc_include_opus) { |
814 defines += [ "WEBRTC_CODEC_OPUS" ] | 789 defines += [ "WEBRTC_CODEC_OPUS" ] |
815 deps += [ ":webrtc_opus" ] | 790 deps += [ ":webrtc_opus" ] |
816 } | 791 } |
817 } | 792 } |
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