Move tests of legacy video code to its own target.
To ensure that tests of non-legacy code doesn't depend on legacy classes and headers. Bug: None Change-Id: Ief63fd77e412892b6f0923530d2317bde4937585 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242364 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35569}
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WebRTC LUCI CQ
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BUILD.gn
1
BUILD.gn
@ -625,6 +625,7 @@ if (rtc_include_tests && !build_with_chromium) {
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"test:test_common",
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"test:test_main",
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"test:video_test_common",
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"video:video_legacy_tests",
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"video:video_tests",
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"video/adaptation:video_adaptation_tests",
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]
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@ -618,7 +618,6 @@ if (rtc_include_tests) {
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"alignment_adjuster_unittest.cc",
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"buffered_frame_decryptor_unittest.cc",
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"call_stats2_unittest.cc",
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"call_stats_unittest.cc",
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"cpu_scaling_tests.cc",
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"encoder_bitrate_adjuster_unittest.cc",
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"encoder_overshoot_detector_unittest.cc",
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@ -649,11 +648,9 @@ if (rtc_include_tests) {
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"quality_scaling_tests.cc",
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"quality_threshold_unittest.cc",
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"receive_statistics_proxy2_unittest.cc",
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"receive_statistics_proxy_unittest.cc",
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"report_block_stats_unittest.cc",
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"rtp_video_stream_receiver2_unittest.cc",
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"rtp_video_stream_receiver_frame_transformer_delegate_unittest.cc",
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"rtp_video_stream_receiver_unittest.cc",
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"send_delay_stats_unittest.cc",
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"send_statistics_proxy_unittest.cc",
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"stats_counter_unittest.cc",
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@ -668,7 +665,6 @@ if (rtc_include_tests) {
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deps = [
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":frame_cadence_adapter",
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":video",
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":video_legacy",
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":video_mocks",
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":video_stream_decoder_impl",
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":video_stream_encoder_impl",
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@ -787,4 +783,37 @@ if (rtc_include_tests) {
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deps += [ "../media:rtc_media_base" ]
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}
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}
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rtc_library("video_legacy_tests") {
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testonly = true
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sources = [
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"call_stats_unittest.cc",
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"receive_statistics_proxy_unittest.cc",
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"rtp_video_stream_receiver_unittest.cc",
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]
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deps = [
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":video_legacy",
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"../api:scoped_refptr",
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"../api/video:video_frame",
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"../api/video:video_frame_type",
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"../api/video:video_rtp_headers",
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"../call:mock_rtp_interfaces",
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"../common_video",
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"../media:rtc_media_base",
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"../modules/rtp_rtcp",
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"../modules/rtp_rtcp:rtp_rtcp_format",
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"../modules/utility",
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"../modules/video_coding",
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"../modules/video_coding:video_codec_interface",
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"../rtc_base:rtc_base_approved",
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"../rtc_base/task_utils:to_queued_task",
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"../system_wrappers",
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"../system_wrappers:field_trial",
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"../system_wrappers:metrics",
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"../test:field_trial",
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"../test:mock_frame_transformer",
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"../test:mock_transport",
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"../test:test_support",
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]
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absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
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}
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}
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@ -60,6 +60,7 @@ class CallStats2Test : public ::testing::Test {
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}
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test::RunLoop loop_;
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// TODO(nisse): Don't use process thread.
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std::unique_ptr<ProcessThread> process_thread_{
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ProcessThread::Create("CallStats")};
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// Note: Since rtc::Thread doesn't support injecting a Clock, we're going
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@ -27,7 +27,6 @@
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#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
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#include "modules/rtp_rtcp/source/rtp_packet_received.h"
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#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
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#include "modules/utility/include/process_thread.h"
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#include "modules/video_coding/frame_object.h"
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#include "modules/video_coding/include/video_coding_defines.h"
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#include "modules/video_coding/rtp_frame_reference_finder.h"
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@ -19,7 +19,6 @@
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#include "api/call/transport.h"
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#include "call/video_receive_stream.h"
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#include "modules/rtp_rtcp/source/rtp_descriptor_authentication.h"
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#include "modules/utility/include/process_thread.h"
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#include "rtc_base/event.h"
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#include "rtc_base/ref_counted_object.h"
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#include "rtc_base/task_utils/to_queued_task.h"
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@ -28,7 +28,6 @@
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#include "media/engine/fake_webrtc_call.h"
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#include "modules/pacing/packet_router.h"
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#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
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#include "modules/utility/include/process_thread.h"
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#include "modules/video_coding/encoded_frame.h"
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#include "rtc_base/event.h"
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#include "system_wrappers/include/clock.h"
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