zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Replace field trials with WebRtcKeyValueConfig in RtpRtcpModule

Browse Source 
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}
This commit is contained in:
Per Kjellander
2019-02-21 07:55:59 +01:00
committed by Commit Bot
parent aa1a43e31f
commit e11b7d2e80
11 changed files with 125 additions and 67 deletions
Download Patch File Download Diff File Expand all files Collapse all files

4
modules/rtp_rtcp/BUILD.gn
View File

@ -202,6 +202,8 @@ rtc_static_library("rtp_rtcp") {
"../../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_static_library("rtp_rtcp") {
"../../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 @@ if (rtc_include_tests) {
"../../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",

2
modules/rtp_rtcp/DEPS
View File

@ -3,4 +3,6 @@ include_rules = [
"+common_video",
"+logging/rtc_event_log",
"+system_wrappers",
# Avoid directly using field_trial. Instead use WebRtcKeyValueConfig.
"-system_wrappers/include/field_trial.h",
]

5
modules/rtp_rtcp/include/rtp_rtcp.h
View File

@ -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 @@ class RtpRtcp : public Module, public RtcpFeedbackSenderInterface {
// 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);
};

11
modules/rtp_rtcp/source/rtp_rtcp_impl.cc
View File

@ -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 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
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 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
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());

7
modules/rtp_rtcp/source/rtp_sender.cc
View File

@ -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 @@ RTPSender::RTPSender(
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 @@ RTPSender::RTPSender(
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.

4
modules/rtp_rtcp/source/rtp_sender.h
View File

@ -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 @@ class RTPSender {
bool populate_network2_timestamp,
FrameEncryptorInterface* frame_encryptor,
bool require_frame_encryption,
bool extmap_allow_mixed);
bool extmap_allow_mixed,
const WebRtcKeyValueConfig& field_trials);
~RTPSender();

4
modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc
View File

@ -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 @@ class RtpSenderAudioTest : public ::testing::Test {
false,
nullptr,
false,
false),
false,
FieldTrialBasedConfig()),
rtp_sender_audio_(&fake_clock_, &rtp_sender_) {
rtp_sender_.SetSSRC(kSsrc);
rtp_sender_.SetSequenceNumber(kSeqNum);

107
modules/rtp_rtcp/source/rtp_sender_unittest.cc
View File

@ -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 @@ class RtpSenderTest : public ::testing::TestWithParam<bool> {
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 @@ TEST_P(RtpSenderTest, AssignSequenceNumberAllowsPaddingOnAudio) {
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 @@ TEST_P(RtpSenderTestWithoutPacer,
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,
}
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, SendsPacketsWithTransportSequenceNumber) {
}
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, DoesnSetIncludedInAllocationByDefault) {
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 @@ TEST_P(RtpSenderTest, SendsPacketsWithTransportSequenceNumber) {
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 @@ TEST_P(RtpSenderTest, OnSendPacketNotUpdatedWithoutSeqNumAllocator) {
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 @@ TEST_P(RtpSenderTest, SendRedundantPayloads) {
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 @@ TEST_P(RtpSenderTestWithoutPacer, SendGenericVideo) {
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 @@ TEST_P(RtpSenderTest, SendFlexfecPackets) {
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 @@ TEST_P(RtpSenderTest, NoFlexfecForTimingFrames) {
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 @@ TEST_P(RtpSenderTestWithoutPacer, SendFlexfecPackets) {
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 @@ TEST_P(RtpSenderTest, FecOverheadRate) {
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 @@ TEST_P(RtpSenderTest, BitrateCallbacks) {
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 @@ TEST_P(RtpSenderTestWithoutPacer, StreamDataCountersCallbacks) {
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 @@ TEST_P(RtpSenderTest, OnOverheadChanged) {
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 @@ TEST_P(RtpSenderTest, DoesNotUpdateOverheadOnEqualSize) {
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, DoesNotUpdateOverheadOnEqualSize) {
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);

7
modules/rtp_rtcp/source/rtp_sender_video.cc
View File

@ -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 @@ RTPSenderVideo::RTPSenderVideo(Clock* clock,
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 @@ RTPSenderVideo::RTPSenderVideo(Clock* clock,
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() {}

3
modules/rtp_rtcp/source/rtp_sender_video.h
View File

@ -54,7 +54,8 @@ class RTPSenderVideo {
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,

38
modules/rtp_rtcp/source/rtp_sender_video_unittest.cc
View File

@ -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 @@ class TestRtpSenderVideo : public RTPSenderVideo {
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 TestRtpSenderVideo : public RTPSenderVideo {
}
};
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 @@ class RtpSenderVideoTest : public ::testing::TestWithParam<bool> {
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 @@ class RtpSenderVideoTest : public ::testing::TestWithParam<bool> {
int version);
protected:
test::ScopedFieldTrials field_trials_;
FieldTrials field_trials_;
SimulatedClock fake_clock_;
LoopbackTransportTest transport_;
RateLimiter retransmission_rate_limiter_;