webrtc/audio/utility/BUILD.gn
Danil Chapovalov f065ff85e2 Cleanup expired field trial WebRTC-VoIPChannelRemixingAdjustmentKillSwitch
Bug: chromium:40108588
Change-Id: Ifc334819dd486ac791b5d04faa6d6bd77a481dd7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349644
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42668}
2024-07-23 13:23:26 +00:00

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1.5 KiB
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# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
import("../../webrtc.gni")
group("utility") {
deps = [ ":audio_frame_operations" ]
}
rtc_library("audio_frame_operations") {
visibility = [ "*" ]
sources = [
"audio_frame_operations.cc",
"audio_frame_operations.h",
"channel_mixer.cc",
"channel_mixer.h",
"channel_mixing_matrix.cc",
"channel_mixing_matrix.h",
]
deps = [
"../../api:array_view",
"../../api/audio:audio_frame_api",
"../../common_audio",
"../../rtc_base:checks",
"../../rtc_base:logging",
"../../rtc_base:safe_conversions",
"//third_party/abseil-cpp/absl/base:core_headers",
]
}
if (rtc_include_tests) {
rtc_library("utility_tests") {
testonly = true
sources = [
"audio_frame_operations_unittest.cc",
"channel_mixer_unittest.cc",
"channel_mixing_matrix_unittest.cc",
]
deps = [
":audio_frame_operations",
"../../api/audio:audio_frame_api",
"../../rtc_base:checks",
"../../rtc_base:logging",
"../../rtc_base:macromagic",
"../../rtc_base:stringutils",
"../../test:test_support",
"//testing/gtest",
]
}
}