We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
99 lines
3.3 KiB
Plaintext
99 lines
3.3 KiB
Plaintext
# Copyright(c) 2020 The WebRTC project authors.All Rights Reserved.
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#
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# Use of this source code is governed by a BSD - style license
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# that can be found in the LICENSE file in the root of the source
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# tree.An additional intellectual property rights grant can be found
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# in the file PATENTS.All contributing project authors may
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# be found in the AUTHORS file in the root of the source tree.
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import("../../../webrtc.gni")
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if (rtc_include_tests) {
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rtc_source_set("mock_task_queue") {
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testonly = true
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visibility = [ "*" ]
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sources = [ "mock_task_queue.h" ]
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deps = [
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"../../../api/task_queue:task_queue",
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"../../../test:test_support",
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]
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}
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if (!build_with_chromium) {
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rtc_library("voip_core_unittests") {
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testonly = true
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sources = [ "voip_core_unittest.cc" ]
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deps = [
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"..:voip_core",
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"../../../api/audio_codecs:builtin_audio_decoder_factory",
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"../../../api/audio_codecs:builtin_audio_encoder_factory",
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"../../../api/task_queue:default_task_queue_factory",
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"../../../modules/audio_device:mock_audio_device",
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"../../../modules/audio_processing:mocks",
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"../../../modules/utility:mock_process_thread",
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"../../../test:audio_codec_mocks",
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"../../../test:mock_transport",
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"../../../test:test_support",
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]
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}
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}
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rtc_library("audio_channel_unittests") {
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testonly = true
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sources = [ "audio_channel_unittest.cc" ]
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deps = [
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":mock_task_queue",
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"..:audio_channel",
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"../../../api:transport_api",
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"../../../api/audio_codecs:builtin_audio_decoder_factory",
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"../../../api/audio_codecs:builtin_audio_encoder_factory",
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"../../../api/task_queue:task_queue",
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"../../../modules/audio_mixer:audio_mixer_impl",
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"../../../modules/audio_mixer:audio_mixer_test_utils",
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"../../../modules/rtp_rtcp:rtp_rtcp",
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"../../../modules/rtp_rtcp:rtp_rtcp_format",
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"../../../modules/utility",
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"../../../rtc_base:logging",
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"../../../test:mock_transport",
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"../../../test:test_support",
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]
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}
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rtc_library("audio_ingress_unittests") {
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testonly = true
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sources = [ "audio_ingress_unittest.cc" ]
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deps = [
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"..:audio_egress",
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"..:audio_ingress",
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"../../../api:transport_api",
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"../../../api/audio_codecs:builtin_audio_decoder_factory",
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"../../../api/audio_codecs:builtin_audio_encoder_factory",
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"../../../api/task_queue:default_task_queue_factory",
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"../../../modules/audio_mixer:audio_mixer_test_utils",
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"../../../modules/rtp_rtcp:rtp_rtcp",
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"../../../rtc_base:logging",
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"../../../rtc_base:rtc_event",
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"../../../test:mock_transport",
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"../../../test:test_support",
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]
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}
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rtc_library("audio_egress_unittests") {
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testonly = true
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sources = [ "audio_egress_unittest.cc" ]
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deps = [
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"..:audio_egress",
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"../../../api:transport_api",
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"../../../api/audio_codecs:builtin_audio_encoder_factory",
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"../../../api/task_queue:default_task_queue_factory",
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"../../../modules/audio_mixer:audio_mixer_test_utils",
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"../../../modules/rtp_rtcp:rtp_rtcp",
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"../../../modules/rtp_rtcp:rtp_rtcp_format",
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"../../../rtc_base:logging",
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"../../../rtc_base:rtc_event",
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"../../../test:mock_transport",
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"../../../test:test_support",
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]
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}
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}
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