Drop RtpRtcp unittest dependency on global field trial string

FieldTrialBasedConfig reads config from the global field trial string
ScopedKeyValueConfig adjust the global field trial string
test::ExplicitKeyValueConfig doesnt touch the global field trial string

Bug: webrtc:11926
Change-Id: I8883634fdc7e1bdb63eec9bf38114a3031103839
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299062
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39683}
This commit is contained in:
Danil Chapovalov 2023-03-24 17:10:31 +01:00 committed by WebRTC LUCI CQ
parent 9a98d0b8e0
commit 7bb9322e9e
9 changed files with 7 additions and 19 deletions

View File

@ -285,7 +285,6 @@ rtc_library("rtp_rtcp") {
"../../api/rtc_event_log",
"../../api/task_queue:pending_task_safety_flag",
"../../api/task_queue:task_queue",
"../../api/transport:field_trial_based_config",
"../../api/transport/rtp:dependency_descriptor",
"../../api/transport/rtp:rtp_source",
"../../api/units:data_rate",
@ -365,7 +364,6 @@ rtc_source_set("rtp_rtcp_legacy") {
"../../api:rtp_headers",
"../../api:transport_api",
"../../api/rtc_event_log",
"../../api/transport:field_trial_based_config",
"../../api/units:data_rate",
"../../api/units:timestamp",
"../../api/video:video_bitrate_allocation",
@ -648,7 +646,6 @@ if (rtc_include_tests) {
"../../api:transport_api",
"../../api/rtc_event_log",
"../../api/task_queue",
"../../api/transport:field_trial_based_config",
"../../api/transport/rtp:dependency_descriptor",
"../../api/units:data_rate",
"../../api/units:data_size",
@ -685,12 +682,10 @@ if (rtc_include_tests) {
"../../rtc_base:timeutils",
"../../system_wrappers",
"../../test:explicit_key_value_config",
"../../test:field_trial",
"../../test:mock_frame_transformer",
"../../test:mock_transport",
"../../test:rtp_test_utils",
"../../test:run_loop",
"../../test:scoped_key_value_config",
"../../test:test_support",
"../../test/time_controller:time_controller",
"../video_coding:codec_globals_headers",

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@ -15,7 +15,6 @@
#include <utility>
#include "absl/strings/match.h"
#include "api/transport/field_trial_based_config.h"
#include "api/units/timestamp.h"
#include "logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h"
#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"

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@ -15,7 +15,6 @@
#include "absl/algorithm/container.h"
#include "api/call/transport.h"
#include "api/transport/field_trial_based_config.h"
#include "call/rtp_stream_receiver_controller.h"
#include "call/rtx_receive_stream.h"
#include "modules/rtp_rtcp/include/receive_statistics.h"
@ -25,6 +24,7 @@
#include "modules/rtp_rtcp/source/rtp_sender_video.h"
#include "rtc_base/rate_limiter.h"
#include "rtc_base/thread.h"
#include "test/explicit_key_value_config.h"
#include "test/gtest.h"
namespace webrtc {
@ -138,7 +138,7 @@ class RtpRtcpRtxNackTest : public ::testing::Test {
configuration.local_media_ssrc = kTestSsrc;
configuration.rtx_send_ssrc = kTestRtxSsrc;
rtp_rtcp_module_ = ModuleRtpRtcpImpl2::Create(configuration);
FieldTrialBasedConfig field_trials;
test::ExplicitKeyValueConfig field_trials("");
RTPSenderVideo::Config video_config;
video_config.clock = &fake_clock;
video_config.rtp_sender = rtp_rtcp_module_->RtpSender();

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@ -21,7 +21,6 @@
#include "absl/strings/string_view.h"
#include "absl/types/optional.h"
#include "api/transport/field_trial_based_config.h"
#include "modules/rtp_rtcp/source/rtcp_packet/dlrr.h"
#include "modules/rtp_rtcp/source/rtcp_sender.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_config.h"

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@ -22,7 +22,6 @@
#include "absl/strings/string_view.h"
#include "absl/types/optional.h"
#include "api/sequence_checker.h"
#include "api/transport/field_trial_based_config.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
#include "modules/rtp_rtcp/source/rtcp_packet/dlrr.h"

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@ -14,13 +14,13 @@
#include <memory>
#include <set>
#include "api/transport/field_trial_based_config.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtcp_packet.h"
#include "modules/rtp_rtcp/source/rtcp_packet/nack.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "modules/rtp_rtcp/source/rtp_sender_video.h"
#include "rtc_base/rate_limiter.h"
#include "test/explicit_key_value_config.h"
#include "test/gmock.h"
#include "test/gtest.h"
#include "test/rtcp_packet_parser.h"
@ -187,7 +187,7 @@ class RtpRtcpImplTest : public ::testing::Test {
sender_.impl_->SetSequenceNumber(kSequenceNumber);
sender_.impl_->SetStorePacketsStatus(true, 100);
FieldTrialBasedConfig field_trials;
test::ExplicitKeyValueConfig field_trials("");
RTPSenderVideo::Config video_config;
video_config.clock = &clock_;
video_config.rtp_sender = sender_.impl_->RtpSender();

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@ -122,8 +122,7 @@ class RtpRtcpInterface : public RtcpFeedbackSenderInterface {
// done by RTCP RR acking.
bool always_send_mid_and_rid = false;
// If set, field trials are read from `field_trials`, otherwise
// defaults to webrtc::FieldTrialBasedConfig.
// If set, field trials are read from `field_trials`.
const FieldTrialsView* field_trials = nullptr;
// SSRCs for media and retransmission, respectively.

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@ -16,7 +16,6 @@
#include <utility>
#include "absl/strings/match.h"
#include "api/transport/field_trial_based_config.h"
#include "logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h"
#include "rtc_base/logging.h"

View File

@ -15,7 +15,6 @@
#include "absl/strings/string_view.h"
#include "api/rtc_event_log/rtc_event.h"
#include "api/transport/field_trial_based_config.h"
#include "api/video/video_codec_constants.h"
#include "api/video/video_timing.h"
#include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
@ -36,11 +35,10 @@
#include "rtc_base/logging.h"
#include "rtc_base/rate_limiter.h"
#include "rtc_base/strings/string_builder.h"
#include "test/field_trial.h"
#include "test/explicit_key_value_config.h"
#include "test/gmock.h"
#include "test/gtest.h"
#include "test/mock_transport.h"
#include "test/scoped_key_value_config.h"
#include "test/time_controller/simulated_time_controller.h"
namespace webrtc {
@ -172,7 +170,7 @@ class RtpSenderTest : public ::testing::Test {
std::unique_ptr<RtpPacketHistory> packet_history_;
std::unique_ptr<RTPSender> rtp_sender_;
const test::ScopedKeyValueConfig field_trials_;
const test::ExplicitKeyValueConfig field_trials_{""};
std::unique_ptr<RtpPacketToSend> BuildRtpPacket(int payload_type,
bool marker_bit,