Replace field trials with WebRtcKeyValueConfig in RtpRtcpModule
Replaces use of field trials in RtpSender and RtpVideoSender with injectable WebRtcKeyValueConfig.
Implementation still defaults to field trials.
BUG: webrtc:10335
Change-Id: I5fc6d327ee5d011ccc41385734b38344df172627
Reviewed-on: https://webrtc-review.googlesource.com/c/123447
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26795}diff --git a/modules/rtp_rtcp/BUILD.gn b/modules/rtp_rtcp/BUILD.gn
index 6f641eb..bf8e69f 100644
--- a/modules/rtp_rtcp/BUILD.gn
+++ b/modules/rtp_rtcp/BUILD.gn
@@ -202,6 +202,8 @@
"../../api:scoped_refptr",
"../../api:transport_api",
"../../api/audio_codecs:audio_codecs_api",
+ "../../api/transport:field_trial_based_config",
+ "../../api/transport:webrtc_key_value_config",
"../../api/video:video_bitrate_allocation",
"../../api/video:video_bitrate_allocator",
"../../api/video:video_frame",
@@ -222,7 +224,6 @@
"../../rtc_base/system:fallthrough",
"../../rtc_base/time:timestamp_extrapolator",
"../../system_wrappers",
- "../../system_wrappers:field_trial",
"../../system_wrappers:metrics",
"../remote_bitrate_estimator",
"../video_coding:codec_globals_headers",
@@ -432,6 +433,7 @@
"../../api:libjingle_peerconnection_api",
"../../api:scoped_refptr",
"../../api:transport_api",
+ "../../api/transport:field_trial_based_config",
"../../api/video:video_bitrate_allocation",
"../../api/video:video_bitrate_allocator",
"../../api/video:video_codec_constants",
diff --git a/modules/rtp_rtcp/DEPS b/modules/rtp_rtcp/DEPS
index 9d9f33c..dac95dd 100644
--- a/modules/rtp_rtcp/DEPS
+++ b/modules/rtp_rtcp/DEPS
@@ -3,4 +3,6 @@
"+common_video",
"+logging/rtc_event_log",
"+system_wrappers",
+ # Avoid directly using field_trial. Instead use WebRtcKeyValueConfig.
+ "-system_wrappers/include/field_trial.h",
]
diff --git a/modules/rtp_rtcp/include/rtp_rtcp.h b/modules/rtp_rtcp/include/rtp_rtcp.h
index 0ad8635..5f7e0d9 100644
--- a/modules/rtp_rtcp/include/rtp_rtcp.h
+++ b/modules/rtp_rtcp/include/rtp_rtcp.h
@@ -18,6 +18,7 @@
#include "absl/strings/string_view.h"
#include "absl/types/optional.h"
+#include "api/transport/webrtc_key_value_config.h"
#include "api/video/video_bitrate_allocation.h"
#include "modules/include/module.h"
#include "modules/rtp_rtcp/include/flexfec_sender.h"
@@ -107,6 +108,10 @@
// Corresponds to extmap-allow-mixed in SDP negotiation.
bool extmap_allow_mixed = false;
+ // If set, field trials are read from |field_trials|, otherwise
+ // defaults to webrtc::FieldTrialBasedConfig.
+ WebRtcKeyValueConfig* field_trials = nullptr;
+
private:
RTC_DISALLOW_COPY_AND_ASSIGN(Configuration);
};
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 3589d6a..3ed38d7 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -18,6 +18,7 @@
#include <utility>
#include "absl/memory/memory.h"
+#include "api/transport/field_trial_based_config.h"
#include "modules/rtp_rtcp/source/rtcp_packet/dlrr.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
#include "rtc_base/checks.h"
@@ -97,6 +98,7 @@
remote_bitrate_(configuration.remote_bitrate_estimator),
rtt_stats_(configuration.rtt_stats),
rtt_ms_(0) {
+ FieldTrialBasedConfig default_trials;
if (!configuration.receiver_only) {
rtp_sender_.reset(new RTPSender(
configuration.audio, configuration.clock,
@@ -113,14 +115,17 @@
configuration.overhead_observer,
configuration.populate_network2_timestamp,
configuration.frame_encryptor, configuration.require_frame_encryption,
- configuration.extmap_allow_mixed));
+ configuration.extmap_allow_mixed,
+ configuration.field_trials ? *configuration.field_trials
+ : default_trials));
if (configuration.audio) {
audio_ = absl::make_unique<RTPSenderAudio>(clock_, rtp_sender_.get());
} else {
video_ = absl::make_unique<RTPSenderVideo>(
clock_, rtp_sender_.get(), configuration.flexfec_sender,
- configuration.frame_encryptor,
- configuration.require_frame_encryption);
+ configuration.frame_encryptor, configuration.require_frame_encryption,
+ configuration.field_trials ? *configuration.field_trials
+ : default_trials);
}
// Make sure rtcp sender use same timestamp offset as rtp sender.
rtcp_sender_.SetTimestampOffset(rtp_sender_->TimestampOffset());
diff --git a/modules/rtp_rtcp/source/rtp_sender.cc b/modules/rtp_rtcp/source/rtp_sender.cc
index 9be7058..197f52b 100644
--- a/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/modules/rtp_rtcp/source/rtp_sender.cc
@@ -32,7 +32,6 @@
#include "rtc_base/numerics/safe_minmax.h"
#include "rtc_base/rate_limiter.h"
#include "rtc_base/time_utils.h"
-#include "system_wrappers/include/field_trial.h"
namespace webrtc {
@@ -104,7 +103,8 @@
bool populate_network2_timestamp,
FrameEncryptorInterface* frame_encryptor,
bool require_frame_encryption,
- bool extmap_allow_mixed)
+ bool extmap_allow_mixed,
+ const WebRtcKeyValueConfig& field_trials)
: clock_(clock),
// TODO(holmer): Remove this conversion?
clock_delta_ms_(clock_->TimeInMilliseconds() - rtc::TimeMillis()),
@@ -148,7 +148,8 @@
overhead_observer_(overhead_observer),
populate_network2_timestamp_(populate_network2_timestamp),
send_side_bwe_with_overhead_(
- webrtc::field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")) {
+ field_trials.Lookup("WebRTC-SendSideBwe-WithOverhead")
+ .find("Enabled") == 0) {
// This random initialization is not intended to be cryptographic strong.
timestamp_offset_ = random_.Rand<uint32_t>();
// Random start, 16 bits. Can't be 0.
diff --git a/modules/rtp_rtcp/source/rtp_sender.h b/modules/rtp_rtcp/source/rtp_sender.h
index 622f41a..f08eb89 100644
--- a/modules/rtp_rtcp/source/rtp_sender.h
+++ b/modules/rtp_rtcp/source/rtp_sender.h
@@ -21,6 +21,7 @@
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/call/transport.h"
+#include "api/transport/webrtc_key_value_config.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/rtp_rtcp/include/flexfec_sender.h"
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
@@ -60,7 +61,8 @@
bool populate_network2_timestamp,
FrameEncryptorInterface* frame_encryptor,
bool require_frame_encryption,
- bool extmap_allow_mixed);
+ bool extmap_allow_mixed,
+ const WebRtcKeyValueConfig& field_trials);
~RTPSender();
diff --git a/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc
index c5ca7f0..31b0048 100644
--- a/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc
@@ -10,6 +10,7 @@
#include <vector>
+#include "api/transport/field_trial_based_config.h"
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
@@ -79,7 +80,8 @@
false,
nullptr,
false,
- false),
+ false,
+ FieldTrialBasedConfig()),
rtp_sender_audio_(&fake_clock_, &rtp_sender_) {
rtp_sender_.SetSSRC(kSsrc);
rtp_sender_.SetSequenceNumber(kSeqNum);
diff --git a/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index 85566c4..2bd3c94 100644
--- a/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -12,6 +12,7 @@
#include <vector>
#include "absl/memory/memory.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/events/rtc_event.h"
@@ -193,7 +194,7 @@
absl::nullopt, &seq_num_allocator_, nullptr, nullptr, nullptr,
&mock_rtc_event_log_, &send_packet_observer_,
&retransmission_rate_limiter_, nullptr, populate_network2, nullptr,
- false, false));
+ false, false, FieldTrialBasedConfig()));
rtp_sender_->SetSequenceNumber(kSeqNum);
rtp_sender_->SetTimestampOffset(0);
rtp_sender_->SetSSRC(kSsrc);
@@ -333,7 +334,7 @@
kEnableAudio, &fake_clock_, &transport, &mock_paced_sender_,
absl::nullopt, nullptr, nullptr, nullptr, nullptr, &mock_rtc_event_log_,
nullptr, &retransmission_rate_limiter_, nullptr, false, nullptr, false,
- false));
+ false, FieldTrialBasedConfig()));
rtp_sender_->SetTimestampOffset(0);
rtp_sender_->SetSSRC(kSsrc);
@@ -376,11 +377,12 @@
TransportFeedbackObserverGetsCorrectByteCount) {
constexpr int kRtpOverheadBytesPerPacket = 12 + 8;
testing::NiceMock<MockOverheadObserver> mock_overhead_observer;
- rtp_sender_.reset(new RTPSender(
- false, &fake_clock_, &transport_, nullptr, absl::nullopt,
- &seq_num_allocator_, &feedback_observer_, nullptr, nullptr,
- &mock_rtc_event_log_, nullptr, &retransmission_rate_limiter_,
- &mock_overhead_observer, false, nullptr, false, false));
+ rtp_sender_.reset(
+ new RTPSender(false, &fake_clock_, &transport_, nullptr, absl::nullopt,
+ &seq_num_allocator_, &feedback_observer_, nullptr, nullptr,
+ &mock_rtc_event_log_, nullptr,
+ &retransmission_rate_limiter_, &mock_overhead_observer,
+ false, nullptr, false, false, FieldTrialBasedConfig()));
rtp_sender_->SetSSRC(kSsrc);
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionTransportSequenceNumber,
@@ -403,11 +405,12 @@
}
TEST_P(RtpSenderTestWithoutPacer, SendsPacketsWithTransportSequenceNumber) {
- rtp_sender_.reset(new RTPSender(
- false, &fake_clock_, &transport_, nullptr, absl::nullopt,
- &seq_num_allocator_, &feedback_observer_, nullptr, nullptr,
- &mock_rtc_event_log_, &send_packet_observer_,
- &retransmission_rate_limiter_, nullptr, false, nullptr, false, false));
+ rtp_sender_.reset(
+ new RTPSender(false, &fake_clock_, &transport_, nullptr, absl::nullopt,
+ &seq_num_allocator_, &feedback_observer_, nullptr, nullptr,
+ &mock_rtc_event_log_, &send_packet_observer_,
+ &retransmission_rate_limiter_, nullptr, false, nullptr,
+ false, false, FieldTrialBasedConfig()));
rtp_sender_->SetSSRC(kSsrc);
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionTransportSequenceNumber,
@@ -435,11 +438,12 @@
}
TEST_P(RtpSenderTestWithoutPacer, PacketOptionsNoRetransmission) {
- rtp_sender_.reset(new RTPSender(
- false, &fake_clock_, &transport_, nullptr, absl::nullopt,
- &seq_num_allocator_, &feedback_observer_, nullptr, nullptr,
- &mock_rtc_event_log_, &send_packet_observer_,
- &retransmission_rate_limiter_, nullptr, false, nullptr, false, false));
+ rtp_sender_.reset(
+ new RTPSender(false, &fake_clock_, &transport_, nullptr, absl::nullopt,
+ &seq_num_allocator_, &feedback_observer_, nullptr, nullptr,
+ &mock_rtc_event_log_, &send_packet_observer_,
+ &retransmission_rate_limiter_, nullptr, false, nullptr,
+ false, false, FieldTrialBasedConfig()));
rtp_sender_->SetSSRC(kSsrc);
SendGenericPacket();
@@ -490,13 +494,14 @@
TEST_P(RtpSenderTestWithoutPacer, OnSendSideDelayUpdated) {
testing::StrictMock<MockSendSideDelayObserver> send_side_delay_observer_;
- rtp_sender_.reset(new RTPSender(
- false, &fake_clock_, &transport_, nullptr, absl::nullopt, nullptr,
- nullptr, nullptr, &send_side_delay_observer_, &mock_rtc_event_log_,
- nullptr, nullptr, nullptr, false, nullptr, false, false));
+ rtp_sender_.reset(
+ new RTPSender(false, &fake_clock_, &transport_, nullptr, absl::nullopt,
+ nullptr, nullptr, nullptr, &send_side_delay_observer_,
+ &mock_rtc_event_log_, nullptr, nullptr, nullptr, false,
+ nullptr, false, false, FieldTrialBasedConfig()));
rtp_sender_->SetSSRC(kSsrc);
RTPSenderVideo rtp_sender_video(&fake_clock_, rtp_sender_.get(), nullptr,
- nullptr, false);
+ nullptr, false, FieldTrialBasedConfig());
const uint8_t kPayloadType = 127;
const char payload_name[] = "GENERIC";
@@ -573,7 +578,8 @@
false, &fake_clock_, &transport_, &mock_paced_sender_, absl::nullopt,
&seq_num_allocator_, &feedback_observer_, nullptr, nullptr,
&mock_rtc_event_log_, &send_packet_observer_,
- &retransmission_rate_limiter_, nullptr, false, nullptr, false, false));
+ &retransmission_rate_limiter_, nullptr, false, nullptr, false, false,
+ FieldTrialBasedConfig()));
rtp_sender_->SetSequenceNumber(kSeqNum);
rtp_sender_->SetSSRC(kSsrc);
rtp_sender_->SetStorePacketsStatus(true, 10);
@@ -958,7 +964,7 @@
false, &fake_clock_, &transport_, &mock_paced_sender_, absl::nullopt,
nullptr /* TransportSequenceNumberAllocator */, nullptr, nullptr, nullptr,
nullptr, &send_packet_observer_, &retransmission_rate_limiter_, nullptr,
- false, nullptr, false, false));
+ false, nullptr, false, false, FieldTrialBasedConfig()));
rtp_sender_->SetSequenceNumber(kSeqNum);
rtp_sender_->SetSSRC(kSsrc);
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
@@ -984,7 +990,8 @@
rtp_sender_.reset(new RTPSender(
false, &fake_clock_, &transport, &mock_paced_sender_, absl::nullopt,
nullptr, nullptr, nullptr, nullptr, &mock_rtc_event_log_, nullptr,
- &retransmission_rate_limiter_, nullptr, false, nullptr, false, false));
+ &retransmission_rate_limiter_, nullptr, false, nullptr, false, false,
+ FieldTrialBasedConfig()));
rtp_sender_->SetSequenceNumber(kSeqNum);
rtp_sender_->SetSSRC(kSsrc);
rtp_sender_->SetRtxPayloadType(kRtxPayload, kPayload);
@@ -1060,7 +1067,7 @@
const char payload_name[] = "GENERIC";
const uint8_t payload_type = 127;
RTPSenderVideo rtp_sender_video(&fake_clock_, rtp_sender_.get(), nullptr,
- nullptr, false);
+ nullptr, false, FieldTrialBasedConfig());
rtp_sender_video.RegisterPayloadType(payload_type, payload_name);
uint8_t payload[] = {47, 11, 32, 93, 89};
@@ -1109,13 +1116,14 @@
false, &fake_clock_, &transport_, &mock_paced_sender_, kFlexfecSsrc,
&seq_num_allocator_, nullptr, nullptr, nullptr, &mock_rtc_event_log_,
&send_packet_observer_, &retransmission_rate_limiter_, nullptr, false,
- nullptr, false, false));
+ nullptr, false, false, FieldTrialBasedConfig()));
rtp_sender_->SetSSRC(kMediaSsrc);
rtp_sender_->SetSequenceNumber(kSeqNum);
rtp_sender_->SetStorePacketsStatus(true, 10);
RTPSenderVideo rtp_sender_video(&fake_clock_, rtp_sender_.get(),
- &flexfec_sender, nullptr, false);
+ &flexfec_sender, nullptr, false,
+ FieldTrialBasedConfig());
rtp_sender_video.RegisterPayloadType(kMediaPayloadType, "GENERIC");
// Parameters selected to generate a single FEC packet per media packet.
@@ -1180,13 +1188,15 @@
false, &fake_clock_, &transport_, &mock_paced_sender_,
flexfec_sender.ssrc(), &seq_num_allocator_, nullptr, nullptr, nullptr,
&mock_rtc_event_log_, &send_packet_observer_,
- &retransmission_rate_limiter_, nullptr, false, nullptr, false, false));
+ &retransmission_rate_limiter_, nullptr, false, nullptr, false, false,
+ FieldTrialBasedConfig()));
rtp_sender_->SetSSRC(kMediaSsrc);
rtp_sender_->SetSequenceNumber(kSeqNum);
rtp_sender_->SetStorePacketsStatus(true, 10);
RTPSenderVideo rtp_sender_video(&fake_clock_, rtp_sender_.get(),
- &flexfec_sender, nullptr, false);
+ &flexfec_sender, nullptr, false,
+ FieldTrialBasedConfig());
rtp_sender_video.RegisterPayloadType(kMediaPayloadType, "GENERIC");
// Need extension to be registered for timing frames to be sent.
@@ -1277,12 +1287,13 @@
false, &fake_clock_, &transport_, nullptr, flexfec_sender.ssrc(),
&seq_num_allocator_, nullptr, nullptr, nullptr, &mock_rtc_event_log_,
&send_packet_observer_, &retransmission_rate_limiter_, nullptr, false,
- nullptr, false, false));
+ nullptr, false, false, FieldTrialBasedConfig()));
rtp_sender_->SetSSRC(kMediaSsrc);
rtp_sender_->SetSequenceNumber(kSeqNum);
RTPSenderVideo rtp_sender_video(&fake_clock_, rtp_sender_.get(),
- &flexfec_sender, nullptr, false);
+ &flexfec_sender, nullptr, false,
+ FieldTrialBasedConfig());
rtp_sender_video.RegisterPayloadType(kMediaPayloadType, "GENERIC");
// Parameters selected to generate a single FEC packet per media packet.
@@ -1404,12 +1415,14 @@
false, &fake_clock_, &transport_, &mock_paced_sender_,
flexfec_sender.ssrc(), &seq_num_allocator_, nullptr, nullptr, nullptr,
&mock_rtc_event_log_, &send_packet_observer_,
- &retransmission_rate_limiter_, nullptr, false, nullptr, false, false));
+ &retransmission_rate_limiter_, nullptr, false, nullptr, false, false,
+ FieldTrialBasedConfig()));
rtp_sender_->SetSSRC(kMediaSsrc);
rtp_sender_->SetSequenceNumber(kSeqNum);
RTPSenderVideo rtp_sender_video(&fake_clock_, rtp_sender_.get(),
- &flexfec_sender, nullptr, false);
+ &flexfec_sender, nullptr, false,
+ FieldTrialBasedConfig());
rtp_sender_video.RegisterPayloadType(kMediaPayloadType, "GENERIC");
// Parameters selected to generate a single FEC packet per media packet.
FecProtectionParams params;
@@ -1469,14 +1482,15 @@
uint32_t total_bitrate_;
uint32_t retransmit_bitrate_;
} callback;
- rtp_sender_.reset(new RTPSender(
- false, &fake_clock_, &transport_, nullptr, absl::nullopt, nullptr,
- nullptr, &callback, nullptr, nullptr, nullptr,
- &retransmission_rate_limiter_, nullptr, false, nullptr, false, false));
+ rtp_sender_.reset(
+ new RTPSender(false, &fake_clock_, &transport_, nullptr, absl::nullopt,
+ nullptr, nullptr, &callback, nullptr, nullptr, nullptr,
+ &retransmission_rate_limiter_, nullptr, false, nullptr,
+ false, false, FieldTrialBasedConfig()));
rtp_sender_->SetSSRC(kSsrc);
RTPSenderVideo rtp_sender_video(&fake_clock_, rtp_sender_.get(), nullptr,
- nullptr, false);
+ nullptr, false, FieldTrialBasedConfig());
const char payload_name[] = "GENERIC";
const uint8_t payload_type = 127;
rtp_sender_video.RegisterPayloadType(payload_type, payload_name);
@@ -1561,7 +1575,7 @@
const char payload_name[] = "GENERIC";
const uint8_t payload_type = 127;
RTPSenderVideo rtp_sender_video(&fake_clock_, rtp_sender_.get(), nullptr,
- nullptr, false);
+ nullptr, false, FieldTrialBasedConfig());
rtp_sender_video.RegisterPayloadType(payload_type, payload_name);
uint8_t payload[] = {47, 11, 32, 93, 89};
rtp_sender_->SetStorePacketsStatus(true, 1);
@@ -1703,7 +1717,7 @@
new RTPSender(false, &fake_clock_, &transport_, nullptr, absl::nullopt,
nullptr, nullptr, nullptr, nullptr, nullptr, nullptr,
&retransmission_rate_limiter_, &mock_overhead_observer,
- false, nullptr, false, false));
+ false, nullptr, false, false, FieldTrialBasedConfig()));
rtp_sender_->SetSSRC(kSsrc);
// RTP overhead is 12B.
@@ -1725,7 +1739,7 @@
new RTPSender(false, &fake_clock_, &transport_, nullptr, absl::nullopt,
nullptr, nullptr, nullptr, nullptr, nullptr, nullptr,
&retransmission_rate_limiter_, &mock_overhead_observer,
- false, nullptr, false, false));
+ false, nullptr, false, false, FieldTrialBasedConfig()));
rtp_sender_->SetSSRC(kSsrc);
EXPECT_CALL(mock_overhead_observer, OnOverheadChanged(_)).Times(1);
@@ -1735,10 +1749,11 @@
TEST_P(RtpSenderTest, SendsKeepAlive) {
MockTransport transport;
- rtp_sender_.reset(new RTPSender(
- false, &fake_clock_, &transport, nullptr, absl::nullopt, nullptr, nullptr,
- nullptr, nullptr, &mock_rtc_event_log_, nullptr,
- &retransmission_rate_limiter_, nullptr, false, nullptr, false, false));
+ rtp_sender_.reset(
+ new RTPSender(false, &fake_clock_, &transport, nullptr, absl::nullopt,
+ nullptr, nullptr, nullptr, nullptr, &mock_rtc_event_log_,
+ nullptr, &retransmission_rate_limiter_, nullptr, false,
+ nullptr, false, false, FieldTrialBasedConfig()));
rtp_sender_->SetSequenceNumber(kSeqNum);
rtp_sender_->SetTimestampOffset(0);
rtp_sender_->SetSSRC(kSsrc);
diff --git a/modules/rtp_rtcp/source/rtp_sender_video.cc b/modules/rtp_rtcp/source/rtp_sender_video.cc
index cc1c58b..32a3621 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_video.cc
@@ -33,7 +33,6 @@
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/trace_event.h"
-#include "system_wrappers/include/field_trial.h"
namespace webrtc {
@@ -188,7 +187,8 @@
RTPSender* rtp_sender,
FlexfecSender* flexfec_sender,
FrameEncryptorInterface* frame_encryptor,
- bool require_frame_encryption)
+ bool require_frame_encryption,
+ const WebRtcKeyValueConfig& field_trials)
: rtp_sender_(rtp_sender),
clock_(clock),
retransmission_settings_(kRetransmitBaseLayer |
@@ -206,7 +206,8 @@
frame_encryptor_(frame_encryptor),
require_frame_encryption_(require_frame_encryption),
generic_descriptor_auth_experiment_(
- field_trial::IsEnabled("WebRTC-GenericDescriptorAuth")) {}
+ field_trials.Lookup("WebRTC-GenericDescriptorAuth").find("Enabled") ==
+ 0) {}
RTPSenderVideo::~RTPSenderVideo() {}
diff --git a/modules/rtp_rtcp/source/rtp_sender_video.h b/modules/rtp_rtcp/source/rtp_sender_video.h
index d29934f..b3c7f30 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video.h
+++ b/modules/rtp_rtcp/source/rtp_sender_video.h
@@ -54,7 +54,8 @@
RTPSender* rtpSender,
FlexfecSender* flexfec_sender,
FrameEncryptorInterface* frame_encryptor,
- bool require_frame_encryption);
+ bool require_frame_encryption,
+ const WebRtcKeyValueConfig& field_trials);
virtual ~RTPSenderVideo();
bool SendVideo(FrameType frame_type,
diff --git a/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc
index 6dd5353..47fabd1 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc
@@ -8,6 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include <string>
#include <vector>
#include "api/video/video_codec_constants.h"
@@ -24,7 +25,6 @@
#include "modules/rtp_rtcp/source/rtp_sender_video.h"
#include "rtc_base/arraysize.h"
#include "rtc_base/rate_limiter.h"
-#include "test/field_trial.h"
#include "test/gmock.h"
#include "test/gtest.h"
@@ -96,8 +96,14 @@
public:
TestRtpSenderVideo(Clock* clock,
RTPSender* rtp_sender,
- FlexfecSender* flexfec_sender)
- : RTPSenderVideo(clock, rtp_sender, flexfec_sender, nullptr, false) {}
+ FlexfecSender* flexfec_sender,
+ const WebRtcKeyValueConfig& field_trials)
+ : RTPSenderVideo(clock,
+ rtp_sender,
+ flexfec_sender,
+ nullptr,
+ false,
+ field_trials) {}
~TestRtpSenderVideo() override {}
StorageType GetStorageType(const RTPVideoHeader& header,
@@ -109,11 +115,26 @@
}
};
+class FieldTrials : public WebRtcKeyValueConfig {
+ public:
+ explicit FieldTrials(bool use_send_side_bwe_with_overhead)
+ : use_send_side_bwe_with_overhead_(use_send_side_bwe_with_overhead) {}
+
+ std::string Lookup(absl::string_view key) const override {
+ return key == "WebRTC-SendSideBwe-WithOverhead" &&
+ use_send_side_bwe_with_overhead_
+ ? "Enabled"
+ : "";
+ }
+
+ private:
+ bool use_send_side_bwe_with_overhead_;
+};
+
class RtpSenderVideoTest : public ::testing::TestWithParam<bool> {
public:
RtpSenderVideoTest()
- : field_trials_(GetParam() ? "WebRTC-SendSideBwe-WithOverhead/Enabled/"
- : ""),
+ : field_trials_(GetParam()),
fake_clock_(kStartTime),
retransmission_rate_limiter_(&fake_clock_, 1000),
// TODO(pbos): Set up to use pacer.
@@ -133,8 +154,9 @@
false,
nullptr,
false,
- false),
- rtp_sender_video_(&fake_clock_, &rtp_sender_, nullptr) {
+ false,
+ field_trials_),
+ rtp_sender_video_(&fake_clock_, &rtp_sender_, nullptr, field_trials_) {
rtp_sender_.SetSequenceNumber(kSeqNum);
rtp_sender_.SetTimestampOffset(0);
rtp_sender_.SetSSRC(kSsrc);
@@ -148,7 +170,7 @@
int version);
protected:
- test::ScopedFieldTrials field_trials_;
+ FieldTrials field_trials_;
SimulatedClock fake_clock_;
LoopbackTransportTest transport_;
RateLimiter retransmission_rate_limiter_;
