xref: /aosp_15_r20/external/webrtc/modules/audio_processing/capture_levels_adjuster/BUILD.gn (revision d9f758449e529ab9291ac668be2861e7a55c2422)
1# Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
2#
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
5# tree. An additional intellectual property rights grant can be found
6# in the file PATENTS.  All contributing project authors may
7# be found in the AUTHORS file in the root of the source tree.
8
9import("../../../webrtc.gni")
10
11rtc_library("capture_levels_adjuster") {
12  visibility = [ "*" ]
13
14  sources = [
15    "audio_samples_scaler.cc",
16    "audio_samples_scaler.h",
17    "capture_levels_adjuster.cc",
18    "capture_levels_adjuster.h",
19  ]
20
21  defines = []
22
23  deps = [
24    "..:audio_buffer",
25    "../../../api:array_view",
26    "../../../rtc_base:checks",
27    "../../../rtc_base:safe_minmax",
28  ]
29}
30
31rtc_library("capture_levels_adjuster_unittests") {
32  testonly = true
33
34  sources = [
35    "audio_samples_scaler_unittest.cc",
36    "capture_levels_adjuster_unittest.cc",
37  ]
38  deps = [
39    ":capture_levels_adjuster",
40    "..:audioproc_test_utils",
41    "../../../rtc_base:gunit_helpers",
42    "../../../rtc_base:stringutils",
43    "../../../test:test_support",
44  ]
45}
46