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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
commit | 6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch) | |
tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/api/voip/BUILD.gn | |
parent | Initial commit. (diff) | |
download | thunderbird-upstream.tar.xz thunderbird-upstream.zip |
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to '')
-rw-r--r-- | third_party/libwebrtc/api/voip/BUILD.gn | 85 |
1 files changed, 85 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/voip/BUILD.gn b/third_party/libwebrtc/api/voip/BUILD.gn new file mode 100644 index 0000000000..714490a526 --- /dev/null +++ b/third_party/libwebrtc/api/voip/BUILD.gn @@ -0,0 +1,85 @@ +# Copyright(c) 2020 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") + +rtc_source_set("voip_api") { + visibility = [ "*" ] + sources = [ + "voip_base.h", + "voip_codec.h", + "voip_dtmf.h", + "voip_engine.h", + "voip_network.h", + "voip_statistics.h", + "voip_volume_control.h", + ] + deps = [ + "..:array_view", + "../audio_codecs:audio_codecs_api", + "../neteq:neteq_api", + ] + absl_deps = [ + "//third_party/abseil-cpp/absl/base:core_headers", + "//third_party/abseil-cpp/absl/types:optional", + ] +} + +rtc_library("voip_engine_factory") { + visibility = [ "*" ] + sources = [ + "voip_engine_factory.cc", + "voip_engine_factory.h", + ] + deps = [ + ":voip_api", + "..:scoped_refptr", + "../../audio/voip:voip_core", + "../../modules/audio_device:audio_device_api", + "../../modules/audio_processing:api", + "../../rtc_base:logging", + "../audio_codecs:audio_codecs_api", + "../task_queue", + ] +} + +if (rtc_include_tests) { + rtc_source_set("mock_voip_engine") { + testonly = true + visibility = [ "*" ] + sources = [ "test/mock_voip_engine.h" ] + deps = [ + ":voip_api", + "..:array_view", + "../../test:test_support", + ] + absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ] + } + + rtc_library("voip_engine_factory_unittests") { + testonly = true + sources = [ "test/voip_engine_factory_unittest.cc" ] + deps = [ + ":voip_engine_factory", + "../../modules/audio_device:mock_audio_device", + "../../modules/audio_processing:mocks", + "../../test:audio_codec_mocks", + "../../test:test_support", + "../task_queue:default_task_queue_factory", + ] + } + + rtc_library("compile_all_headers") { + testonly = true + sources = [ "test/compile_all_headers.cc" ] + deps = [ + ":mock_voip_engine", + "../../test:test_support", + ] + } +} |