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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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783 "../../system_wrappers", | 813 "../../system_wrappers", |
784 ] | 814 ] |
785 | 815 |
786 defines = [] | 816 defines = [] |
787 | 817 |
788 if (rtc_include_opus) { | 818 if (rtc_include_opus) { |
789 defines += [ "WEBRTC_CODEC_OPUS" ] | 819 defines += [ "WEBRTC_CODEC_OPUS" ] |
790 deps += [ ":webrtc_opus" ] | 820 deps += [ ":webrtc_opus" ] |
791 } | 821 } |
792 } | 822 } |
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