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Cleanup merge differences from upstream

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Co-authored-by: Rashad Sookram <rashad@signal.org>
This commit is contained in:
Jim Gustafson 2023-01-18 16:59:06 -08:00
parent fe9009be92
commit 67b9dd5a62
34 changed files with 85 additions and 146 deletions
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1
BUILD.gn
View file

@ -651,7 +651,6 @@ if (rtc_include_tests && !build_with_chromium) {
"test:test_common",
"test:test_main",
"test:video_test_common",
"video:video_legacy_tests",
"video:video_tests",
"video/adaptation:video_adaptation_tests",
]

7
api/DEPS
View file

@ -248,13 +248,6 @@ specific_include_rules = {
"+rtc_base/ref_counted_object.h",
],
"fake_metronome\.h": [
"+rtc_base/synchronization/mutex.h",
"+rtc_base/task_queue.h",
"+rtc_base/task_utils/repeating_task.h",
"+rtc_base/thread_annotations.h",
],
"mock.*\.h": [
"+test/gmock.h",
],

17
api/metronome/BUILD.gn
View file

@ -16,21 +16,4 @@ rtc_source_set("metronome") {
"../task_queue",
"../units:time_delta",
]
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
if (rtc_include_tests) {
rtc_library("timestamp_extrapolator_unittests") {
testonly = true
sources = [ "timestamp_extrapolator_unittest.cc" ]
deps = [
":timestamp_extrapolator",
"../../api/units:frequency",
"../../api/units:time_delta",
"../../api/units:timestamp",
"../../system_wrappers",
"../../test:test_support",
]
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
}

5
audio/channel_receive.cc
View file

@ -713,9 +713,8 @@ void ChannelReceive::ReceivePacket(const uint8_t* packet,
payload = decrypted_audio_payload.data();
payload_data_length = decrypted_audio_payload.size();
} else if (crypto_options_.sframe.require_frame_encryption) {
// RingRTC change to avoid spurious log.
// RTC_DLOG(LS_ERROR)
// << "FrameDecryptor required but not set, dropping packet";
RTC_DLOG(LS_ERROR)
<< "FrameDecryptor required but not set, dropping packet";
payload_data_length = 0;
}

1
call/bitrate_allocator.cc
View file

@ -380,7 +380,6 @@ void BitrateAllocator::OnNetworkEstimateChanged(TargetTransferRate msg) {
RTC_DCHECK_RUN_ON(&sequenced_checker_);
last_target_bps_ = msg.target_rate.bps();
last_stable_target_bps_ = msg.stable_target_rate.bps();
last_non_zero_bitrate_bps_ =
last_target_bps_ > 0 ? last_target_bps_ : last_non_zero_bitrate_bps_;

11
call/rtp_demuxer.cc
View file

@ -113,12 +113,11 @@ std::string RtpDemuxer::DescribePacket(const RtpPacketReceived& packet) {
RtpDemuxer::RtpDemuxer(bool use_mid /* = true*/) : use_mid_(use_mid) {}
RtpDemuxer::~RtpDemuxer() {
// RingRTC change to avoid spurious crashes
// RTC_DCHECK(sink_by_mid_.empty());
// RTC_DCHECK(sink_by_ssrc_.empty());
// RTC_DCHECK(sinks_by_pt_.empty());
// RTC_DCHECK(sink_by_mid_and_rsid_.empty());
// RTC_DCHECK(sink_by_rsid_.empty());
RTC_DCHECK(sink_by_mid_.empty());
RTC_DCHECK(sink_by_ssrc_.empty());
RTC_DCHECK(sinks_by_pt_.empty());
RTC_DCHECK(sink_by_mid_and_rsid_.empty());
RTC_DCHECK(sink_by_rsid_.empty());
}
bool RtpDemuxer::AddSink(const RtpDemuxerCriteria& criteria,

16
media/engine/webrtc_video_engine_unittest.cc
View file

@ -5556,7 +5556,7 @@ TEST_F(WebRtcVideoChannelTest, GetAggregatedStatsReportWithoutSubStreams) {
EXPECT_EQ(sender.total_encoded_bytes_target,
stats.total_encoded_bytes_target);
// Comes from substream only.
EXPECT_EQ(sender.total_packet_send_delay_ms, 0u);
EXPECT_EQ(sender.total_packet_send_delay, webrtc::TimeDelta::Zero());
EXPECT_EQ(sender.qp_sum, absl::nullopt);
EXPECT_EQ(sender.has_entered_low_resolution,
@ -5583,7 +5583,8 @@ TEST_F(WebRtcVideoChannelTest, GetAggregatedStatsReportForSubStreams) {
substream.retransmit_bitrate_bps = 6;
substream.avg_delay_ms = 7;
substream.max_delay_ms = 8;
substream.total_packet_send_delay_ms = 9;
substream.rtp_stats.transmitted.total_packet_delay =
webrtc::TimeDelta::Millis(9);
substream.rtp_stats.transmitted.header_bytes = 10;
substream.rtp_stats.transmitted.padding_bytes = 11;
substream.rtp_stats.retransmitted.payload_bytes = 12;
@ -5629,6 +5630,8 @@ TEST_F(WebRtcVideoChannelTest, GetAggregatedStatsReportForSubStreams) {
static_cast<int>(2 * substream.rtp_stats.transmitted.packets));
EXPECT_EQ(sender.retransmitted_packets_sent,
2u * substream.rtp_stats.retransmitted.packets);
EXPECT_EQ(sender.total_packet_send_delay,
2 * substream.rtp_stats.transmitted.total_packet_delay);
EXPECT_EQ(sender.packets_lost,
2 * substream.report_block_data->report_block().packets_lost);
EXPECT_EQ(sender.fraction_lost,
@ -5678,8 +5681,6 @@ TEST_F(WebRtcVideoChannelTest, GetAggregatedStatsReportForSubStreams) {
EXPECT_EQ(sender.total_encode_time_ms, 2u * substream.total_encode_time_ms);
EXPECT_EQ(sender.total_encoded_bytes_target,
2u * substream.total_encoded_bytes_target);
EXPECT_EQ(sender.total_packet_send_delay_ms,
2u * substream.total_packet_send_delay_ms);
EXPECT_EQ(sender.has_entered_low_resolution,
stats.has_entered_low_resolution);
EXPECT_EQ(sender.qp_sum, 2u * *substream.qp_sum);
@ -5705,7 +5706,8 @@ TEST_F(WebRtcVideoChannelTest, GetPerLayerStatsReportForSubStreams) {
substream.retransmit_bitrate_bps = 6;
substream.avg_delay_ms = 7;
substream.max_delay_ms = 8;
substream.total_packet_send_delay_ms = 9;
substream.rtp_stats.transmitted.total_packet_delay =
webrtc::TimeDelta::Millis(9);
substream.rtp_stats.transmitted.header_bytes = 10;
substream.rtp_stats.transmitted.padding_bytes = 11;
substream.rtp_stats.retransmitted.payload_bytes = 12;
@ -5749,6 +5751,8 @@ TEST_F(WebRtcVideoChannelTest, GetPerLayerStatsReportForSubStreams) {
substream.rtp_stats.retransmitted.payload_bytes);
EXPECT_EQ(sender.packets_sent,
static_cast<int>(substream.rtp_stats.transmitted.packets));
EXPECT_EQ(sender.total_packet_send_delay,
substream.rtp_stats.transmitted.total_packet_delay);
EXPECT_EQ(sender.retransmitted_packets_sent,
substream.rtp_stats.retransmitted.packets);
EXPECT_EQ(sender.packets_lost,
@ -5799,8 +5803,6 @@ TEST_F(WebRtcVideoChannelTest, GetPerLayerStatsReportForSubStreams) {
EXPECT_EQ(sender.total_encode_time_ms, substream.total_encode_time_ms);
EXPECT_EQ(sender.total_encoded_bytes_target,
substream.total_encoded_bytes_target);
EXPECT_EQ(sender.total_packet_send_delay_ms,
substream.total_packet_send_delay_ms);
EXPECT_EQ(sender.has_entered_low_resolution,
stats.has_entered_low_resolution);
EXPECT_EQ(sender.qp_sum, *substream.qp_sum);

10
media/engine/webrtc_voice_engine.cc
View file

@ -585,12 +585,10 @@ WebRtcVoiceEngine::GetRtpHeaderExtensions() const {
std::vector<webrtc::RtpHeaderExtensionCapability> result;
int id = 1;
// RingRTC change to disable unused header extensions
for (const auto& uri :
{// webrtc::RtpExtension::kAudioLevelUri,
webrtc::RtpExtension::kAbsSendTimeUri,
webrtc::RtpExtension::kTransportSequenceNumberUri,
webrtc::RtpExtension::kMidUri,
}) {
for (const auto& uri : {// webrtc::RtpExtension::kAudioLevelUri,
webrtc::RtpExtension::kAbsSendTimeUri,
webrtc::RtpExtension::kTransportSequenceNumberUri,
webrtc::RtpExtension::kMidUri}) {
result.emplace_back(uri, id++, webrtc::RtpTransceiverDirection::kSendRecv);
}
return result;

6
media/engine/webrtc_voice_engine_unittest.cc
View file

@ -1019,10 +1019,8 @@ TEST_P(WebRtcVoiceEngineTestFake, ChangeRecvCodecPayloadType) {
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
}
// Test that we enable Opus/Red with the field trial.
TEST_P(WebRtcVoiceEngineTestFake, RecvRed) {
webrtc::test::ScopedFieldTrials override_field_trials(
"WebRTC-Audio-Red-For-Opus/Enabled/");
// Test that we do allow setting Opus/Red by default.
TEST_P(WebRtcVoiceEngineTestFake, RecvRedDefault) {
EXPECT_TRUE(SetupRecvStream());
cricket::AudioRecvParameters parameters;
parameters.codecs.push_back(kOpusCodec);

1
modules/audio_device/audio_device_impl.cc
View file

@ -664,7 +664,6 @@ int32_t AudioDeviceModuleImpl::MinMicrophoneVolume(uint32_t* minVolume) const {
return 0;
}
// RingRTC changes to add some useful logging
int16_t AudioDeviceModuleImpl::PlayoutDevices() {
RTC_LOG(LS_INFO) << __FUNCTION__;
CHECKinitialized_();

3
modules/audio_processing/aec3/matched_filter.h
View file

@ -89,9 +89,6 @@ void MatchedFilterCore(size_t x_start_index,
// Find largest peak of squared values in array.
size_t MaxSquarePeakIndex(rtc::ArrayView<const float> h);
// Find largest peak of squared values in array.
size_t MaxSquarePeakIndex(rtc::ArrayView<const float> h);
} // namespace aec3
// Produces recursively updated cross-correlation estimates for several signal

1
modules/congestion_controller/goog_cc/robust_throughput_estimator.h
View file

@ -15,7 +15,6 @@
#include <vector>
#include "absl/types/optional.h"
#include "api/field_trials_view.h"
#include "api/transport/network_types.h"
#include "api/units/data_rate.h"
#include "api/units/timestamp.h"

10
modules/congestion_controller/goog_cc/send_side_bandwidth_estimation_unittest.cc
View file

@ -10,7 +10,6 @@
#include "modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.h"
#include "api/network_state_predictor.h"
#include "api/rtc_event_log/rtc_event.h"
#include "logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.h"
#include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
@ -55,8 +54,7 @@ void TestProbing(bool use_delay_based) {
// Initial REMB applies immediately.
if (use_delay_based) {
bwe.UpdateDelayBasedEstimate(Timestamp::Millis(now_ms),
DataRate::BitsPerSec(kRembBps),
BandwidthUsage::kBwNormal);
DataRate::BitsPerSec(kRembBps));
} else {
bwe.UpdateReceiverEstimate(Timestamp::Millis(now_ms),
DataRate::BitsPerSec(kRembBps));
@ -68,8 +66,7 @@ void TestProbing(bool use_delay_based) {
now_ms += 2001;
if (use_delay_based) {
bwe.UpdateDelayBasedEstimate(Timestamp::Millis(now_ms),
DataRate::BitsPerSec(kSecondRembBps),
BandwidthUsage::kBwNormal);
DataRate::BitsPerSec(kSecondRembBps));
} else {
bwe.UpdateReceiverEstimate(Timestamp::Millis(now_ms),
DataRate::BitsPerSec(kSecondRembBps));
@ -160,8 +157,7 @@ TEST(SendSideBweTest, SettingSendBitrateOverridesDelayBasedEstimate) {
Timestamp::Millis(now_ms));
bwe.UpdateDelayBasedEstimate(Timestamp::Millis(now_ms),
DataRate::BitsPerSec(kDelayBasedBitrateBps),
BandwidthUsage::kBwNormal);
DataRate::BitsPerSec(kDelayBasedBitrateBps));
bwe.UpdateEstimate(Timestamp::Millis(now_ms));
EXPECT_GE(bwe.target_rate().bps(), kInitialBitrateBps);
EXPECT_LE(bwe.target_rate().bps(), kDelayBasedBitrateBps);

10
modules/desktop_capture/desktop_frame.cc
View file

@ -45,13 +45,9 @@ void DesktopFrame::CopyPixelsFrom(const uint8_t* src_buffer,
RTC_CHECK(DesktopRect::MakeSize(size()).ContainsRect(dest_rect));
uint8_t* dest = GetFrameDataAtPos(dest_rect.top_left());
// TODO(crbug.com/1330019): Temporary workaround for a known libyuv crash when
// the height or width is 0. Remove this once this change has been merged.
if (dest_rect.width() && dest_rect.height()) {
libyuv::CopyPlane(src_buffer, src_stride, dest, stride(),
DesktopFrame::kBytesPerPixel * dest_rect.width(),
dest_rect.height());
}
libyuv::CopyPlane(src_buffer, src_stride, dest, stride(),
DesktopFrame::kBytesPerPixel * dest_rect.width(),
dest_rect.height());
}
void DesktopFrame::CopyPixelsFrom(const DesktopFrame& src_frame,

8
modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.cc
View file

@ -90,7 +90,6 @@ DEPRECATED_RtpSenderEgress::DEPRECATED_RtpSenderEgress(
timestamp_offset_(0),
max_delay_it_(send_delays_.end()),
sum_delays_ms_(0),
total_packet_send_delay_ms_(0),
send_rates_(kNumMediaTypes,
{kBitrateStatisticsWindowMs, RateStatistics::kBpsScale}),
rtp_sequence_number_map_(need_rtp_packet_infos_
@ -341,7 +340,6 @@ void DEPRECATED_RtpSenderEgress::UpdateDelayStatistics(int64_t capture_time_ms,
int avg_delay_ms = 0;
int max_delay_ms = 0;
uint64_t total_packet_send_delay_ms = 0;
{
MutexLock lock(&lock_);
// Compute the max and average of the recent capture-to-send delays.
@ -392,8 +390,6 @@ void DEPRECATED_RtpSenderEgress::UpdateDelayStatistics(int64_t capture_time_ms,
max_delay_it_ = it;
}
sum_delays_ms_ += new_send_delay;
total_packet_send_delay_ms_ += new_send_delay;
total_packet_send_delay_ms = total_packet_send_delay_ms_;
size_t num_delays = send_delays_.size();
RTC_DCHECK(max_delay_it_ != send_delays_.end());
@ -405,8 +401,8 @@ void DEPRECATED_RtpSenderEgress::UpdateDelayStatistics(int64_t capture_time_ms,
avg_delay_ms =
rtc::dchecked_cast<int>((sum_delays_ms_ + num_delays / 2) / num_delays);
}
send_side_delay_observer_->SendSideDelayUpdated(
avg_delay_ms, max_delay_ms, total_packet_send_delay_ms, ssrc);
send_side_delay_observer_->SendSideDelayUpdated(avg_delay_ms, max_delay_ms,
ssrc);
}
void DEPRECATED_RtpSenderEgress::RecomputeMaxSendDelay() {

1
modules/rtp_rtcp/source/rtp_packet_history.h
View file

@ -18,7 +18,6 @@
#include <utility>
#include <vector>
#include "absl/base/attributes.h"
#include "api/function_view.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"

5
modules/video_coding/svc/create_scalability_structure.h
View file

@ -30,11 +30,6 @@ std::unique_ptr<ScalableVideoController> CreateScalabilityStructure(
absl::optional<ScalableVideoController::StreamLayersConfig>
ScalabilityStructureConfig(ScalabilityMode name);
// Returns descrption of the scalability structure identified by 'name',
// Return nullopt for unknown name.
absl::optional<ScalableVideoController::StreamLayersConfig>
ScalabilityStructureConfig(absl::string_view name);
} // namespace webrtc
#endif // MODULES_VIDEO_CODING_SVC_CREATE_SCALABILITY_STRUCTURE_H_

2
net/dcsctp/socket/dcsctp_socket.h
View file

@ -292,8 +292,6 @@ class DcSctpSocket : public DcSctpSocketInterface {
State state_ = State::kClosed;
// If the connection is established, contains a transmission control block.
std::unique_ptr<TransmissionControlBlock> tcb_;
SctpImplementation peer_implementation_ = SctpImplementation::kUnknown;
};
} // namespace dcsctp

13
net/dcsctp/socket/state_cookie_test.cc
View file

@ -55,18 +55,5 @@ TEST(StateCookieTest, ValidateMagicValue) {
EXPECT_EQ(magic, "dcSCTP00");
}
TEST(StateCookieTest, ValidateMagicValue) {
Capabilities capabilities = {/*partial_reliability=*/true,
/*message_interleaving=*/false,
/*reconfig=*/true};
StateCookie cookie(VerificationTag(123), TSN(456),
/*a_rwnd=*/789, TieTag(101112), capabilities);
std::vector<uint8_t> serialized = cookie.Serialize();
ASSERT_THAT(serialized, SizeIs(StateCookie::kCookieSize));
absl::string_view magic(reinterpret_cast<const char*>(serialized.data()), 8);
EXPECT_EQ(magic, "dcSCTP00");
}
} // namespace
} // namespace dcsctp

6
p2p/base/fake_port_allocator.h
View file

@ -17,6 +17,7 @@
#include <vector>
#include "absl/strings/string_view.h"
#include "api/make_ref_counted.h"
#include "p2p/base/basic_packet_socket_factory.h"
#include "p2p/base/ice_credentials_iterator.h"
#include "p2p/base/ice_gatherer.h"
@ -218,7 +219,6 @@ class FakePortAllocatorSession : public PortAllocatorSession {
class FakePortAllocator : public cricket::PortAllocator {
public:
// TODO(bugs.webrtc.org/13145): Require non-null `factory`.
FakePortAllocator(rtc::Thread* network_thread,
rtc::PacketSocketFactory* factory)
: FakePortAllocator(network_thread, factory, nullptr) {}
@ -236,7 +236,7 @@ class FakePortAllocator : public cricket::PortAllocator {
absl::string_view ice_pwd) override {
return new FakePortAllocatorSession(
this, network_thread_, factory_.get(), std::string(content_name),
component, std::string(ice_ufrag), std::string(ice_pwd, field_trials_);
component, std::string(ice_ufrag), std::string(ice_pwd), field_trials_);
}
rtc::scoped_refptr<webrtc::IceGathererInterface> CreateIceGatherer(
@ -248,7 +248,7 @@ class FakePortAllocator : public cricket::PortAllocator {
auto session = new_allocator->CreateSession(
content_name, 1, parameters.ufrag, parameters.pwd);
return rtc::make_ref_counted<cricket::BasicIceGatherer>(
network_thread_, std::move(new_allocator), std::move(session)), field_trials_);
network_thread_, std::move(new_allocator), std::move(session));
}
bool initialized() const { return initialized_; }

10
p2p/base/p2p_transport_channel_unittest.cc
View file

@ -1282,16 +1282,6 @@ const P2PTransportChannelMatrixTest::Result*
LTRT},
};
class P2PTransportChannelTestWithFieldTrials
: public P2PTransportChannelTest,
public ::testing::WithParamInterface<std::string> {
public:
void Test(const Result& expected) override {
webrtc::test::ScopedFieldTrials field_trials(GetParam());
P2PTransportChannelTest::Test(expected);
}
};
// The actual tests that exercise all the various configurations.
// Test names are of the form P2PTransportChannelTest_TestOPENToNAT_FULL_CONE
#define P2P_TEST_DECLARATION(x, y, z) \

4
p2p/base/port.cc
View file

@ -101,8 +101,10 @@ std::string Port::ComputeFoundation(absl::string_view type,
absl::string_view protocol,
absl::string_view relay_protocol,
const rtc::SocketAddress& base_address) {
// TODO(bugs.webrtc.org/14605): ensure IceTiebreaker() is set.
rtc::StringBuilder sb;
sb << type << base_address.ipaddr().ToString() << protocol << relay_protocol;
sb << type << base_address.ipaddr().ToString() << protocol << relay_protocol
<< rtc::ToString(IceTiebreaker());
return rtc::ToString(rtc::ComputeCrc32(sb.Release()));
}

3
p2p/base/port_allocator.h
View file

@ -100,6 +100,9 @@ enum class IceRegatheringReason {
MAX_VALUE
};
// RingRTC change to avoid incorrect use of flag (enable for test builds)
//const uint32_t kDefaultPortAllocatorFlags = 0;
const uint32_t kDefaultStepDelay = 1000; // 1 sec step delay.
// As per RFC 5245 Appendix B.1, STUN transactions need to be paced at certain
// internal. Less than 20ms is not acceptable. We choose 50ms as our default.

4
pc/media_session.cc
View file

@ -1657,7 +1657,7 @@ std::unique_ptr<SessionDescription> MediaSessionDescriptionFactory::CreateOffer(
session_options.media_description_options.size());
}
// RingRTC change to configure OPUS
// RingRTC change to add ICE forking
IceCredentialsIterator ice_credentials(session_options.ice_credentials);
std::vector<const ContentInfo*> current_active_contents;
@ -1795,7 +1795,7 @@ MediaSessionDescriptionFactory::CreateAnswer(
RTC_DCHECK_EQ(offer->contents().size(),
session_options.media_description_options.size());
// RingRTC change to configure OPUS
// RingRTC change to add ICE forking
IceCredentialsIterator ice_credentials(session_options.ice_credentials);
std::vector<const ContentInfo*> current_active_contents;

1
pc/peer_connection_integrationtest.cc
View file

@ -21,7 +21,6 @@
#include <memory>
#include <string>
#include <tuple>
#include <type_traits>
#include <utility>
#include <vector>

40
pc/rtc_stats_collector_unittest.cc
View file

@ -38,6 +38,7 @@
#include "api/video/video_frame.h"
#include "api/video/video_sink_interface.h"
#include "api/video/video_source_interface.h"
#include "api/video/video_timing.h"
#include "common_video/include/quality_limitation_reason.h"
#include "media/base/media_channel.h"
#include "modules/audio_processing/include/audio_processing_statistics.h"
@ -2703,6 +2704,42 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Video) {
EXPECT_TRUE(report->Get(*expected_video.codec_id));
}
TEST_F(RTCStatsCollectorTest, CollectGoogTimingFrameInfo) {
cricket::VideoMediaInfo video_media_info;
video_media_info.receivers.push_back(cricket::VideoReceiverInfo());
video_media_info.receivers[0].local_stats.push_back(
cricket::SsrcReceiverInfo());
video_media_info.receivers[0].local_stats[0].ssrc = 1;
TimingFrameInfo timing_frame_info;
timing_frame_info.rtp_timestamp = 1;
timing_frame_info.capture_time_ms = 2;
timing_frame_info.encode_start_ms = 3;
timing_frame_info.encode_finish_ms = 4;
timing_frame_info.packetization_finish_ms = 5;
timing_frame_info.pacer_exit_ms = 6;
timing_frame_info.network_timestamp_ms = 7;
timing_frame_info.network2_timestamp_ms = 8;
timing_frame_info.receive_start_ms = 9;
timing_frame_info.receive_finish_ms = 10;
timing_frame_info.decode_start_ms = 11;
timing_frame_info.decode_finish_ms = 12;
timing_frame_info.render_time_ms = 13;
timing_frame_info.flags = 14;
video_media_info.receivers[0].timing_frame_info = timing_frame_info;
pc_->AddVideoChannel("Mid0", "Transport0", video_media_info);
stats_->SetupRemoteTrackAndReceiver(
cricket::MEDIA_TYPE_VIDEO, "RemoteVideoTrackID", "RemoteStreamId", 1);
rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
auto inbound_rtps = report->GetStatsOfType<RTCInboundRTPStreamStats>();
ASSERT_EQ(inbound_rtps.size(), 1u);
ASSERT_TRUE(inbound_rtps[0]->goog_timing_frame_info.is_defined());
EXPECT_EQ(*inbound_rtps[0]->goog_timing_frame_info,
"1,2,3,4,5,6,7,8,9,10,11,12,13,1,0");
}
TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Audio) {
cricket::VoiceMediaInfo voice_media_info;
@ -2787,7 +2824,8 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Video) {
video_media_info.senders[0].key_frames_encoded = 3;
video_media_info.senders[0].total_encode_time_ms = 9000;
video_media_info.senders[0].total_encoded_bytes_target = 1234;
video_media_info.senders[0].total_packet_send_delay_ms = 10000;
video_media_info.senders[0].total_packet_send_delay =
webrtc::TimeDelta::Seconds(10);
video_media_info.senders[0].quality_limitation_reason =
QualityLimitationReason::kBandwidth;
video_media_info.senders[0].quality_limitation_durations_ms

2
pc/rtc_stats_integrationtest.cc
View file

@ -936,6 +936,7 @@ class RTCStatsReportVerifier {
inbound_stream.total_interruption_duration);
verifier.TestMemberIsNonNegative<double>(
inbound_stream.min_playout_delay);
verifier.TestMemberIsDefined(inbound_stream.goog_timing_frame_info);
} else {
verifier.TestMemberIsUndefined(inbound_stream.frames_decoded);
verifier.TestMemberIsUndefined(inbound_stream.key_frames_decoded);
@ -964,6 +965,7 @@ class RTCStatsReportVerifier {
verifier.TestMemberIsNonNegative<double>(
inbound_stream.total_interruption_duration);
verifier.TestMemberIsUndefined(inbound_stream.min_playout_delay);
verifier.TestMemberIsUndefined(inbound_stream.goog_timing_frame_info);
}
return verifier.ExpectAllMembersSuccessfullyTested();
}

1
rtc_base/event_tracer.cc
View file

@ -204,6 +204,7 @@ class EventLogger final {
RTC_CHECK(g_event_logging_active.compare_exchange_strong(zero, 1));
// Finally start, everything should be set up now.
// RingRTC change to keep the tracing thread priority low
logging_thread_ =
PlatformThread::SpawnJoinable([this] { Log(); }, "EventTracingThread",
ThreadAttributes().SetPriority(ThreadPriority::kLow));

8
rtc_base/network.cc
View file

@ -1001,14 +1001,6 @@ void BasicNetworkManager::StopNetworkMonitor() {
IPAddress BasicNetworkManager::QueryDefaultLocalAddress(int family) const {
RTC_DCHECK(family == AF_INET || family == AF_INET6);
// TODO(bugs.webrtc.org/13145): Delete support for null `socket_factory_`,
// require socket factory to be provided to constructor.
SocketFactory* socket_factory = socket_factory_;
if (!socket_factory) {
socket_factory = thread_->socketserver();
}
RTC_DCHECK(socket_factory);
std::unique_ptr<Socket> socket(
socket_factory_->CreateSocket(family, SOCK_DGRAM));
if (!socket) {

2
rtc_base/physical_socket_server.cc
View file

@ -750,8 +750,6 @@ bool SocketDispatcher::IsDescriptorClosed() {
return true;
// The normal blocking error; don't log anything.
case EWOULDBLOCK:
// Interrupted system call.
case EINTR:
return false;
default:
// Assume that all other errors are just blocking errors, meaning the

3
sdk/android/OWNERS
View file

@ -7,6 +7,3 @@ ssilkin@webrtc.org
magjed@webrtc.org
xalep@webrtc.org
per-file ...Audio*.java=henrika@webrtc.org
per-file PRESUBMIT.py=jansson@webrtc.org
per-file PRESUBMIT.py=jleconte@webrtc.org

5
sdk/android/tests/src/org/webrtc/FakeMediaCodecWrapper.java
View file

@ -306,11 +306,6 @@ public class FakeMediaCodecWrapper implements MediaCodecWrapper {
return inputFormat;
}
@Override
public MediaFormat getInputFormat() {
return inputFormat;
}
@Override
public MediaFormat getOutputFormat() {
return outputFormat;

2
test/fuzzers/vp9_encoder_references_fuzzer.cc
View file

@ -34,8 +34,6 @@ using test::FuzzDataHelper;
constexpr int kBitrateEnabledBps = 100'000;
constexpr int kBitrateEnabledBps = 100'000;
class FrameValidator : public EncodedImageCallback {
public:
~FrameValidator() override = default;

12
video/send_statistics_proxy_unittest.cc
View file

@ -322,26 +322,18 @@ TEST_F(SendStatisticsProxyTest, SendSideDelay) {
// stream.
int avg_delay_ms = ssrc;
int max_delay_ms = ssrc + 1;
uint64_t total_packet_send_delay_ms = ssrc + 2;
observer->SendSideDelayUpdated(avg_delay_ms, max_delay_ms,
total_packet_send_delay_ms, ssrc);
observer->SendSideDelayUpdated(avg_delay_ms, max_delay_ms, ssrc);
expected_.substreams[ssrc].avg_delay_ms = avg_delay_ms;
expected_.substreams[ssrc].max_delay_ms = max_delay_ms;
expected_.substreams[ssrc].total_packet_send_delay_ms =
total_packet_send_delay_ms;
}
for (const auto& ssrc : config_.rtp.rtx.ssrcs) {
// Use ssrc as avg_delay_ms and max_delay_ms to get a unique value for each
// stream.
int avg_delay_ms = ssrc;
int max_delay_ms = ssrc + 1;
uint64_t total_packet_send_delay_ms = ssrc + 2;
observer->SendSideDelayUpdated(avg_delay_ms, max_delay_ms,
total_packet_send_delay_ms, ssrc);
observer->SendSideDelayUpdated(avg_delay_ms, max_delay_ms, ssrc);
expected_.substreams[ssrc].avg_delay_ms = avg_delay_ms;
expected_.substreams[ssrc].max_delay_ms = max_delay_ms;
expected_.substreams[ssrc].total_packet_send_delay_ms =
total_packet_send_delay_ms;
}
VideoSendStream::Stats stats = statistics_proxy_->GetStats();
ExpectEqual(expected_, stats);