021eef32f4
Reason for revert: Try again. Original issue's description: > Revert of Refactor webrtc_perf_tests into several source_sets. (patchset #5 id:100001 of https://codereview.webrtc.org/2609403002/ ) > > Reason for revert: > Probably cause unexpected performance regression > BUG=chromium:678569 > > Original issue's description: > > Refactor webrtc_perf_tests into several source_sets. > > > > BUG=webrtc:6954 > > > > Review-Url: https://codereview.webrtc.org/2609403002 > > Cr-Commit-Position: refs/heads/master@{#15902} > > Committed:0b5a26a576> > TBR=kjellander@webrtc.org,ehmaldonado@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2613913002 > Cr-Commit-Position: refs/heads/master@{#15916} > Committed:5fbcd228f0TBR=kjellander@webrtc.org,danilchap@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=chromium:678569 Review-Url: https://codereview.webrtc.org/2615873002 Cr-Commit-Position: refs/heads/master@{#15919}
96 lines
2.5 KiB
Plaintext
96 lines
2.5 KiB
Plaintext
# Copyright (c) 2015 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("../build/webrtc.gni")
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rtc_source_set("call_interfaces") {
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sources = [
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"audio_receive_stream.h",
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"audio_send_stream.cc",
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"audio_send_stream.h",
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"audio_state.h",
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"call.h",
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"flexfec_receive_stream.h",
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]
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}
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rtc_static_library("call") {
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sources = [
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"bitrate_allocator.cc",
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"call.cc",
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"flexfec_receive_stream_impl.cc",
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"flexfec_receive_stream_impl.h",
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]
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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public_deps = [
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":call_interfaces",
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"../api:call_api",
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]
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deps = [
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":call_interfaces",
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"..:webrtc_common",
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"../api:transport_api",
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"../audio",
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"../base:rtc_task_queue",
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"../logging:rtc_event_log_impl",
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"../modules/congestion_controller",
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"../modules/rtp_rtcp",
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"../system_wrappers",
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"../video",
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]
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}
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if (rtc_include_tests) {
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rtc_source_set("call_tests") {
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testonly = true
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sources = [
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"bitrate_allocator_unittest.cc",
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"bitrate_estimator_tests.cc",
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"call_unittest.cc",
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"flexfec_receive_stream_unittest.cc",
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"packet_injection_tests.cc",
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]
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deps = [
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":call",
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"../base:rtc_base_approved",
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"../modules/audio_device:mock_audio_device",
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"../modules/audio_mixer",
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"../test:test_common",
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"//testing/gmock",
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"//testing/gtest",
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]
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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}
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rtc_source_set("call_perf_tests") {
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testonly = true
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sources = [
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"call_perf_tests.cc",
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"rampup_tests.cc",
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"rampup_tests.h",
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]
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deps = [
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"//testing/gtest",
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"//webrtc/test:test_common",
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]
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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}
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}
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