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Bug 1766646 - Vendor libwebrtc from afb246b5a9 

Upstream commit: https://webrtc.googlesource.com/src/+/afb246b5a949d7bf24dd93c3c3900bbebd407224
    Update pc/ to not use implicit conversion from scoped_refptr<T> to T*.

    Bug: webrtc:13464
    Change-Id: I768646af8ded6338ef51486b8d69db1ad71e9a2c
    Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259500
    Reviewed-by: Henrik Boström <hbos@webrtc.org>
    Reviewed-by: Harald Alvestrand <hta@webrtc.org>
    Commit-Queue: Niels Moller <nisse@webrtc.org>
    Cr-Commit-Position: refs/heads/main@{#36588}
This commit is contained in:
Michael Froman 2022-07-13 13:52:04 -05:00 коммит произвёл Connor Sheehan
Родитель 974a4efd7d
Коммит c9f49934e6
45 изменённых файлов: 485 добавлений и 431 удалений
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3
third_party/libwebrtc/README.moz-ff-commit поставляемый
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@ -13302,3 +13302,6 @@ ba1657c95c
# MOZ_LIBWEBRTC_SRC=/home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src MOZ_LIBWEBRTC_COMMIT=mjfdev bash dom/media/webrtc/third_party_build/fast-forward-libwebrtc.sh
# base of lastest vendoring
46501b9192
# MOZ_LIBWEBRTC_SRC=/home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src MOZ_LIBWEBRTC_COMMIT=mjfdev bash dom/media/webrtc/third_party_build/fast-forward-libwebrtc.sh
# base of lastest vendoring
afb246b5a9

2
third_party/libwebrtc/README.mozilla поставляемый
Просмотреть файл

@ -8880,3 +8880,5 @@ libwebrtc updated from /home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwe
libwebrtc updated from /home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src commit mjfdev on 2022-07-13T18:50:40.410459.
# python3 vendor-libwebrtc.py --from-local /home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src --commit mjfdev libwebrtc
libwebrtc updated from /home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src commit mjfdev on 2022-07-13T18:51:21.021217.
# python3 vendor-libwebrtc.py --from-local /home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src --commit mjfdev libwebrtc
libwebrtc updated from /home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src commit mjfdev on 2022-07-13T18:51:59.914692.

1
third_party/libwebrtc/api/scoped_refptr.h поставляемый
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@ -104,6 +104,7 @@ class scoped_refptr {
}
T* get() const { return ptr_; }
explicit operator bool() const { return ptr_ != nullptr; }
operator T*() const { return ptr_; }
T& operator*() const { return *ptr_; }
T* operator->() const { return ptr_; }

4
third_party/libwebrtc/pc/audio_rtp_receiver.cc поставляемый
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@ -251,7 +251,7 @@ void AudioRtpReceiver::SetStreams(
}
}
if (removed) {
existing_stream->RemoveTrack(track_);
existing_stream->RemoveTrack(track_.get());
}
}
// Add remote track to any streams that are new.
@ -265,7 +265,7 @@ void AudioRtpReceiver::SetStreams(
}
}
if (added) {
stream->AddTrack(track_);
stream->AddTrack(track_.get());
}
}
streams_ = streams;

2
third_party/libwebrtc/pc/jsep_transport.cc поставляемый
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@ -202,7 +202,7 @@ webrtc::RTCError JsepTransport::SetLocalJsepTransportDescription(
if (!local_fp) {
local_certificate_ = nullptr;
} else {
error = VerifyCertificateFingerprint(local_certificate_, local_fp);
error = VerifyCertificateFingerprint(local_certificate_.get(), local_fp);
if (!error.ok()) {
local_description_.reset();
return error;

8
third_party/libwebrtc/pc/media_stream_observer.cc поставляемый
Просмотреть файл

@ -54,7 +54,7 @@ void MediaStreamObserver::OnChanged() {
[cached_track](const AudioTrackVector::value_type& new_track) {
return new_track->id() == cached_track->id();
})) {
audio_track_removed_callback_(cached_track.get(), stream_);
audio_track_removed_callback_(cached_track.get(), stream_.get());
}
}
@ -65,7 +65,7 @@ void MediaStreamObserver::OnChanged() {
[new_track](const AudioTrackVector::value_type& cached_track) {
return new_track->id() == cached_track->id();
})) {
audio_track_added_callback_(new_track.get(), stream_);
audio_track_added_callback_(new_track.get(), stream_.get());
}
}
@ -76,7 +76,7 @@ void MediaStreamObserver::OnChanged() {
[cached_track](const VideoTrackVector::value_type& new_track) {
return new_track->id() == cached_track->id();
})) {
video_track_removed_callback_(cached_track.get(), stream_);
video_track_removed_callback_(cached_track.get(), stream_.get());
}
}
@ -87,7 +87,7 @@ void MediaStreamObserver::OnChanged() {
[new_track](const VideoTrackVector::value_type& cached_track) {
return new_track->id() == cached_track->id();
})) {
video_track_added_callback_(new_track.get(), stream_);
video_track_added_callback_(new_track.get(), stream_.get());
}
}

2
third_party/libwebrtc/pc/media_stream_observer.h поставляемый
Просмотреть файл

@ -34,7 +34,7 @@ class MediaStreamObserver : public ObserverInterface {
video_track_removed_callback);
~MediaStreamObserver() override;
const MediaStreamInterface* stream() const { return stream_; }
const MediaStreamInterface* stream() const { return stream_.get(); }
void OnChanged() override;

18
third_party/libwebrtc/pc/media_stream_unittest.cc поставляемый
Просмотреть файл

@ -66,10 +66,10 @@ class MediaStreamTest : public ::testing::Test {
ASSERT_TRUE(audio_track_.get() != NULL);
EXPECT_EQ(MediaStreamTrackInterface::kLive, audio_track_->state());
EXPECT_TRUE(stream_->AddTrack(video_track_));
EXPECT_FALSE(stream_->AddTrack(video_track_));
EXPECT_TRUE(stream_->AddTrack(audio_track_));
EXPECT_FALSE(stream_->AddTrack(audio_track_));
EXPECT_TRUE(stream_->AddTrack(video_track_.get()));
EXPECT_FALSE(stream_->AddTrack(video_track_.get()));
EXPECT_TRUE(stream_->AddTrack(audio_track_.get()));
EXPECT_FALSE(stream_->AddTrack(audio_track_.get()));
}
void ChangeTrack(MediaStreamTrackInterface* track) {
@ -118,17 +118,17 @@ TEST_F(MediaStreamTest, GetTrackInfo) {
}
TEST_F(MediaStreamTest, RemoveTrack) {
MockObserver observer(stream_);
MockObserver observer(stream_.get());
EXPECT_CALL(observer, OnChanged()).Times(Exactly(2));
EXPECT_TRUE(stream_->RemoveTrack(audio_track_));
EXPECT_FALSE(stream_->RemoveTrack(audio_track_));
EXPECT_TRUE(stream_->RemoveTrack(audio_track_.get()));
EXPECT_FALSE(stream_->RemoveTrack(audio_track_.get()));
EXPECT_EQ(0u, stream_->GetAudioTracks().size());
EXPECT_EQ(0u, stream_->GetAudioTracks().size());
EXPECT_TRUE(stream_->RemoveTrack(video_track_));
EXPECT_FALSE(stream_->RemoveTrack(video_track_));
EXPECT_TRUE(stream_->RemoveTrack(video_track_.get()));
EXPECT_FALSE(stream_->RemoveTrack(video_track_.get()));
EXPECT_EQ(0u, stream_->GetVideoTracks().size());
EXPECT_EQ(0u, stream_->GetVideoTracks().size());

10
third_party/libwebrtc/pc/peer_connection.cc поставляемый
Просмотреть файл

@ -644,7 +644,7 @@ RTCError PeerConnection::Initialize(
stats_collector_ = RTCStatsCollector::Create(this);
sdp_handler_ = SdpOfferAnswerHandler::Create(this, configuration,
dependencies, context_);
dependencies, context_.get());
rtp_manager_ = std::make_unique<RtpTransmissionManager>(
IsUnifiedPlan(), signaling_thread(), worker_thread(), channel_manager(),
@ -858,7 +858,7 @@ RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>> PeerConnection::AddTrack(
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_STATE,
"PeerConnection is closed.");
}
if (rtp_manager()->FindSenderForTrack(track)) {
if (rtp_manager()->FindSenderForTrack(track.get())) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INVALID_PARAMETER,
"Sender already exists for track " + track->id() + ".");
@ -866,7 +866,7 @@ RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>> PeerConnection::AddTrack(
auto sender_or_error = rtp_manager()->AddTrack(track, stream_ids);
if (sender_or_error.ok()) {
sdp_handler_->UpdateNegotiationNeeded();
stats_->AddTrack(track);
stats_->AddTrack(track.get());
}
return sender_or_error;
}
@ -898,11 +898,11 @@ RTCError PeerConnection::RemoveTrackOrError(
bool removed;
if (sender->media_type() == cricket::MEDIA_TYPE_AUDIO) {
removed = rtp_manager()->GetAudioTransceiver()->internal()->RemoveSender(
sender);
sender.get());
} else {
RTC_DCHECK_EQ(cricket::MEDIA_TYPE_VIDEO, sender->media_type());
removed = rtp_manager()->GetVideoTransceiver()->internal()->RemoveSender(
sender);
sender.get());
}
if (!removed) {
LOG_AND_RETURN_ERROR(

2
third_party/libwebrtc/pc/peer_connection_adaptation_integrationtest.cc поставляемый
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@ -65,7 +65,7 @@ TrackWithPeriodicSource CreateTrackWithPeriodicSource(
periodic_track_source_config, /* remote */ false);
TrackWithPeriodicSource track_with_source;
track_with_source.track =
factory->CreateVideoTrack("PeriodicTrack", periodic_track_source);
factory->CreateVideoTrack("PeriodicTrack", periodic_track_source.get());
track_with_source.periodic_track_source = periodic_track_source;
return track_with_source;
}

6
third_party/libwebrtc/pc/peer_connection_crypto_unittest.cc поставляемый
Просмотреть файл

@ -110,7 +110,7 @@ class PeerConnectionCryptoBaseTest : public ::testing::Test {
return nullptr;
}
observer->SetPeerConnectionInterface(result.value());
observer->SetPeerConnectionInterface(result.value().get());
return std::make_unique<PeerConnectionWrapper>(
pc_factory_, result.MoveValue(), std::move(observer));
}
@ -662,10 +662,10 @@ TEST_P(PeerConnectionCryptoDtlsCertGenTest, TestCertificateGeneration) {
rtc::make_ref_counted<MockCreateSessionDescriptionObserver>();
observers.push_back(observer);
if (sdp_type_ == SdpType::kOffer) {
pc->pc()->CreateOffer(observer,
pc->pc()->CreateOffer(observer.get(),
PeerConnectionInterface::RTCOfferAnswerOptions());
} else {
pc->pc()->CreateAnswer(observer,
pc->pc()->CreateAnswer(observer.get(),
PeerConnectionInterface::RTCOfferAnswerOptions());
}
}

2
third_party/libwebrtc/pc/peer_connection_data_channel_unittest.cc поставляемый
Просмотреть файл

@ -144,7 +144,7 @@ class PeerConnectionDataChannelBaseTest : public ::testing::Test {
return nullptr;
}
observer->SetPeerConnectionInterface(result.value());
observer->SetPeerConnectionInterface(result.value().get());
auto wrapper = std::make_unique<PeerConnectionWrapperForDataChannelTest>(
pc_factory, result.MoveValue(), std::move(observer));
wrapper->set_sctp_transport_factory(fake_sctp_transport_factory);

80
third_party/libwebrtc/pc/peer_connection_end_to_end_unittest.cc поставляемый
Просмотреть файл

@ -509,14 +509,16 @@ TEST_P(PeerConnectionEndToEndTest, CreateDataChannelBeforeNegotiate) {
Negotiate();
WaitForConnection();
WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 0);
WaitForDataChannelsToOpen(callee_dc, caller_signaled_data_channels_, 0);
WaitForDataChannelsToOpen(caller_dc.get(), callee_signaled_data_channels_, 0);
WaitForDataChannelsToOpen(callee_dc.get(), caller_signaled_data_channels_, 0);
TestDataChannelSendAndReceive(caller_dc, callee_signaled_data_channels_[0]);
TestDataChannelSendAndReceive(callee_dc, caller_signaled_data_channels_[0]);
TestDataChannelSendAndReceive(caller_dc.get(),
callee_signaled_data_channels_[0].get());
TestDataChannelSendAndReceive(callee_dc.get(),
caller_signaled_data_channels_[0].get());
CloseDataChannels(caller_dc, callee_signaled_data_channels_, 0);
CloseDataChannels(callee_dc, caller_signaled_data_channels_, 0);
CloseDataChannels(caller_dc.get(), callee_signaled_data_channels_, 0);
CloseDataChannels(callee_dc.get(), caller_signaled_data_channels_, 0);
}
// Verifies that a DataChannel created after the negotiation can transition to
@ -534,7 +536,7 @@ TEST_P(PeerConnectionEndToEndTest, CreateDataChannelAfterNegotiate) {
WaitForConnection();
// Wait for the data channel created pre-negotiation to be opened.
WaitForDataChannelsToOpen(dummy, callee_signaled_data_channels_, 0);
WaitForDataChannelsToOpen(dummy.get(), callee_signaled_data_channels_, 0);
// Create new DataChannels after the negotiation and verify their states.
rtc::scoped_refptr<DataChannelInterface> caller_dc(
@ -542,14 +544,16 @@ TEST_P(PeerConnectionEndToEndTest, CreateDataChannelAfterNegotiate) {
rtc::scoped_refptr<DataChannelInterface> callee_dc(
callee_->CreateDataChannel("hello", init));
WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 1);
WaitForDataChannelsToOpen(callee_dc, caller_signaled_data_channels_, 0);
WaitForDataChannelsToOpen(caller_dc.get(), callee_signaled_data_channels_, 1);
WaitForDataChannelsToOpen(callee_dc.get(), caller_signaled_data_channels_, 0);
TestDataChannelSendAndReceive(caller_dc, callee_signaled_data_channels_[1]);
TestDataChannelSendAndReceive(callee_dc, caller_signaled_data_channels_[0]);
TestDataChannelSendAndReceive(caller_dc.get(),
callee_signaled_data_channels_[1].get());
TestDataChannelSendAndReceive(callee_dc.get(),
caller_signaled_data_channels_[0].get());
CloseDataChannels(caller_dc, callee_signaled_data_channels_, 1);
CloseDataChannels(callee_dc, caller_signaled_data_channels_, 0);
CloseDataChannels(caller_dc.get(), callee_signaled_data_channels_, 1);
CloseDataChannels(callee_dc.get(), caller_signaled_data_channels_, 0);
}
// Verifies that a DataChannel created can transfer large messages.
@ -566,7 +570,7 @@ TEST_P(PeerConnectionEndToEndTest, CreateDataChannelLargeTransfer) {
WaitForConnection();
// Wait for the data channel created pre-negotiation to be opened.
WaitForDataChannelsToOpen(dummy, callee_signaled_data_channels_, 0);
WaitForDataChannelsToOpen(dummy.get(), callee_signaled_data_channels_, 0);
// Create new DataChannels after the negotiation and verify their states.
rtc::scoped_refptr<DataChannelInterface> caller_dc(
@ -574,16 +578,16 @@ TEST_P(PeerConnectionEndToEndTest, CreateDataChannelLargeTransfer) {
rtc::scoped_refptr<DataChannelInterface> callee_dc(
callee_->CreateDataChannel("hello", init));
WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 1);
WaitForDataChannelsToOpen(callee_dc, caller_signaled_data_channels_, 0);
WaitForDataChannelsToOpen(caller_dc.get(), callee_signaled_data_channels_, 1);
WaitForDataChannelsToOpen(callee_dc.get(), caller_signaled_data_channels_, 0);
TestDataChannelSendAndReceive(caller_dc, callee_signaled_data_channels_[1],
256 * 1024);
TestDataChannelSendAndReceive(callee_dc, caller_signaled_data_channels_[0],
256 * 1024);
TestDataChannelSendAndReceive(
caller_dc.get(), callee_signaled_data_channels_[1].get(), 256 * 1024);
TestDataChannelSendAndReceive(
callee_dc.get(), caller_signaled_data_channels_[0].get(), 256 * 1024);
CloseDataChannels(caller_dc, callee_signaled_data_channels_, 1);
CloseDataChannels(callee_dc, caller_signaled_data_channels_, 0);
CloseDataChannels(caller_dc.get(), callee_signaled_data_channels_, 1);
CloseDataChannels(callee_dc.get(), caller_signaled_data_channels_, 0);
}
// Verifies that DataChannel IDs are even/odd based on the DTLS roles.
@ -628,14 +632,18 @@ TEST_P(PeerConnectionEndToEndTest,
Negotiate();
WaitForConnection();
WaitForDataChannelsToOpen(caller_dc_1, callee_signaled_data_channels_, 0);
WaitForDataChannelsToOpen(caller_dc_2, callee_signaled_data_channels_, 1);
WaitForDataChannelsToOpen(caller_dc_1.get(), callee_signaled_data_channels_,
0);
WaitForDataChannelsToOpen(caller_dc_2.get(), callee_signaled_data_channels_,
1);
std::unique_ptr<webrtc::MockDataChannelObserver> dc_1_observer(
new webrtc::MockDataChannelObserver(callee_signaled_data_channels_[0]));
new webrtc::MockDataChannelObserver(
callee_signaled_data_channels_[0].get()));
std::unique_ptr<webrtc::MockDataChannelObserver> dc_2_observer(
new webrtc::MockDataChannelObserver(callee_signaled_data_channels_[1]));
new webrtc::MockDataChannelObserver(
callee_signaled_data_channels_[1].get()));
const std::string message_1 = "hello 1";
const std::string message_2 = "hello 2";
@ -666,7 +674,7 @@ TEST_P(PeerConnectionEndToEndTest,
Negotiate();
WaitForConnection();
WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 0);
WaitForDataChannelsToOpen(caller_dc.get(), callee_signaled_data_channels_, 0);
int first_channel_id = caller_dc->id();
// Wait for the local side to say it's closed, but not the remote side.
// Previously, the channel on which Close is called reported being closed
@ -676,13 +684,14 @@ TEST_P(PeerConnectionEndToEndTest,
// Create a new channel and ensure it works after closing the previous one.
caller_dc = caller_->CreateDataChannel("data2", init);
WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 1);
WaitForDataChannelsToOpen(caller_dc.get(), callee_signaled_data_channels_, 1);
// Since the second channel was created after the first finished closing, it
// should be able to re-use the first one's ID.
EXPECT_EQ(first_channel_id, caller_dc->id());
TestDataChannelSendAndReceive(caller_dc, callee_signaled_data_channels_[1]);
TestDataChannelSendAndReceive(caller_dc.get(),
callee_signaled_data_channels_[1].get());
CloseDataChannels(caller_dc, callee_signaled_data_channels_, 1);
CloseDataChannels(caller_dc.get(), callee_signaled_data_channels_, 1);
}
// Similar to the above test, but don't wait for the first channel to finish
@ -699,20 +708,21 @@ TEST_P(PeerConnectionEndToEndTest,
Negotiate();
WaitForConnection();
WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 0);
WaitForDataChannelsToOpen(caller_dc.get(), callee_signaled_data_channels_, 0);
int first_channel_id = caller_dc->id();
caller_dc->Close();
// Immediately create a new channel, before waiting for the previous one to
// transition to "closed".
caller_dc = caller_->CreateDataChannel("data2", init);
WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 1);
WaitForDataChannelsToOpen(caller_dc.get(), callee_signaled_data_channels_, 1);
// Since the second channel was created while the first was still closing,
// it should have been assigned a different ID.
EXPECT_NE(first_channel_id, caller_dc->id());
TestDataChannelSendAndReceive(caller_dc, callee_signaled_data_channels_[1]);
TestDataChannelSendAndReceive(caller_dc.get(),
callee_signaled_data_channels_[1].get());
CloseDataChannels(caller_dc, callee_signaled_data_channels_, 1);
CloseDataChannels(caller_dc.get(), callee_signaled_data_channels_, 1);
}
// This tests that if a data channel is closed remotely while not referenced
@ -730,7 +740,7 @@ TEST_P(PeerConnectionEndToEndTest, CloseDataChannelRemotelyWhileNotReferenced) {
Negotiate();
WaitForConnection();
WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 0);
WaitForDataChannelsToOpen(caller_dc.get(), callee_signaled_data_channels_, 0);
// This removes the reference to the remote data channel that we hold.
callee_signaled_data_channels_.clear();
caller_dc->Close();

4
third_party/libwebrtc/pc/peer_connection_factory_unittest.cc поставляемый
Просмотреть файл

@ -472,9 +472,9 @@ TEST_F(PeerConnectionFactoryTest, LocalRendering) {
ASSERT_TRUE(source.get() != NULL);
rtc::scoped_refptr<VideoTrackInterface> track(
factory_->CreateVideoTrack("testlabel", source));
factory_->CreateVideoTrack("testlabel", source.get()));
ASSERT_TRUE(track.get() != NULL);
FakeVideoTrackRenderer local_renderer(track);
FakeVideoTrackRenderer local_renderer(track.get());
EXPECT_EQ(0, local_renderer.num_rendered_frames());
source->InjectFrame(frame_source.GetFrame());

2
third_party/libwebrtc/pc/peer_connection_header_extension_unittest.cc поставляемый
Просмотреть файл

@ -106,7 +106,7 @@ class PeerConnectionHeaderExtensionTest
auto result = pc_factory->CreatePeerConnectionOrError(
config, std::move(pc_dependencies));
EXPECT_TRUE(result.ok());
observer->SetPeerConnectionInterface(result.value());
observer->SetPeerConnectionInterface(result.value().get());
return std::make_unique<PeerConnectionWrapper>(
pc_factory, result.MoveValue(), std::move(observer));
}

2
third_party/libwebrtc/pc/peer_connection_histogram_unittest.cc поставляемый
Просмотреть файл

@ -352,7 +352,7 @@ class PeerConnectionUsageHistogramTest : public ::testing::Test {
return nullptr;
}
observer->SetPeerConnectionInterface(result.value());
observer->SetPeerConnectionInterface(result.value().get());
auto wrapper = std::make_unique<PeerConnectionWrapperForUsageHistogramTest>(
pc_factory, result.MoveValue(), std::move(observer));
return wrapper;

6
third_party/libwebrtc/pc/peer_connection_ice_unittest.cc поставляемый
Просмотреть файл

@ -175,7 +175,7 @@ class PeerConnectionIceBaseTest : public ::testing::Test {
return nullptr;
}
observer->SetPeerConnectionInterface(result.value());
observer->SetPeerConnectionInterface(result.value().get());
auto wrapper = std::make_unique<PeerConnectionWrapperForIceTest>(
pc_factory_, result.MoveValue(), std::move(observer));
wrapper->set_network(fake_network);
@ -809,7 +809,7 @@ TEST_P(PeerConnectionIceTest,
// Chain an operation that will block AddIceCandidate() from executing.
auto answer_observer =
rtc::make_ref_counted<MockCreateSessionDescriptionObserver>();
callee->pc()->CreateAnswer(answer_observer, RTCOfferAnswerOptions());
callee->pc()->CreateAnswer(answer_observer.get(), RTCOfferAnswerOptions());
auto jsep_candidate =
callee->CreateJsepCandidateForFirstTransport(&candidate);
@ -857,7 +857,7 @@ TEST_P(PeerConnectionIceTest,
// Chain an operation that will block AddIceCandidate() from executing.
auto answer_observer =
rtc::make_ref_counted<MockCreateSessionDescriptionObserver>();
callee->pc()->CreateAnswer(answer_observer, RTCOfferAnswerOptions());
callee->pc()->CreateAnswer(answer_observer.get(), RTCOfferAnswerOptions());
auto jsep_candidate =
callee->CreateJsepCandidateForFirstTransport(&candidate);

51
third_party/libwebrtc/pc/peer_connection_integrationtest.cc поставляемый
Просмотреть файл

@ -578,7 +578,8 @@ TEST_P(PeerConnectionIntegrationTest, AudioToVideoUpgrade) {
ASSERT_EQ(2U, transceivers.size());
ASSERT_EQ(cricket::MEDIA_TYPE_VIDEO,
transceivers[1]->receiver()->media_type());
transceivers[1]->sender()->SetTrack(caller()->CreateLocalVideoTrack());
transceivers[1]->sender()->SetTrack(
caller()->CreateLocalVideoTrack().get());
transceivers[1]->SetDirectionWithError(
RtpTransceiverDirection::kSendRecv);
});
@ -1333,8 +1334,9 @@ TEST_P(PeerConnectionIntegrationTest, GetBytesReceivedStatsWithOldStatsApi) {
for (const auto& receiver : callee()->pc()->GetReceivers()) {
// We received frames, so we definitely should have nonzero "received bytes"
// stats at this point.
EXPECT_GT(callee()->OldGetStatsForTrack(receiver->track())->BytesReceived(),
0);
EXPECT_GT(
callee()->OldGetStatsForTrack(receiver->track().get())->BytesReceived(),
0);
}
}
@ -1355,8 +1357,8 @@ TEST_P(PeerConnectionIntegrationTest, GetBytesSentStatsWithOldStatsApi) {
// The callee received frames, so we definitely should have nonzero "sent
// bytes" stats at this point.
EXPECT_GT(caller()->OldGetStatsForTrack(audio_track)->BytesSent(), 0);
EXPECT_GT(caller()->OldGetStatsForTrack(video_track)->BytesSent(), 0);
EXPECT_GT(caller()->OldGetStatsForTrack(audio_track.get())->BytesSent(), 0);
EXPECT_GT(caller()->OldGetStatsForTrack(video_track.get())->BytesSent(), 0);
}
// Test that we can get capture start ntp time.
@ -1379,10 +1381,9 @@ TEST_P(PeerConnectionIntegrationTest, GetCaptureStartNtpTimeWithOldStatsApi) {
// Get the audio output level stats. Note that the level is not available
// until an RTCP packet has been received.
EXPECT_TRUE_WAIT(
callee()->OldGetStatsForTrack(remote_audio_track)->CaptureStartNtpTime() >
0,
2 * kMaxWaitForFramesMs);
EXPECT_TRUE_WAIT(callee()->OldGetStatsForTrack(remote_audio_track.get())
->CaptureStartNtpTime() > 0,
2 * kMaxWaitForFramesMs);
}
// Test that the track ID is associated with all local and remote SSRC stats
@ -2434,10 +2435,14 @@ TEST_F(PeerConnectionIntegrationTestPlanB,
EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
callee()->ice_connection_state(), kMaxWaitForFramesMs);
// Now set the tracks, and expect frames to immediately start flowing.
EXPECT_TRUE(caller_audio_sender->SetTrack(caller()->CreateLocalAudioTrack()));
EXPECT_TRUE(caller_video_sender->SetTrack(caller()->CreateLocalVideoTrack()));
EXPECT_TRUE(callee_audio_sender->SetTrack(callee()->CreateLocalAudioTrack()));
EXPECT_TRUE(callee_video_sender->SetTrack(callee()->CreateLocalVideoTrack()));
EXPECT_TRUE(
caller_audio_sender->SetTrack(caller()->CreateLocalAudioTrack().get()));
EXPECT_TRUE(
caller_video_sender->SetTrack(caller()->CreateLocalVideoTrack().get()));
EXPECT_TRUE(
callee_audio_sender->SetTrack(callee()->CreateLocalAudioTrack().get()));
EXPECT_TRUE(
callee_video_sender->SetTrack(callee()->CreateLocalVideoTrack().get()));
MediaExpectations media_expectations;
media_expectations.ExpectBidirectionalAudioAndVideo();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
@ -2473,10 +2478,14 @@ TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
// Now set the tracks, and expect frames to immediately start flowing.
auto callee_audio_sender = callee()->pc()->GetSenders()[0];
auto callee_video_sender = callee()->pc()->GetSenders()[1];
ASSERT_TRUE(caller_audio_sender->SetTrack(caller()->CreateLocalAudioTrack()));
ASSERT_TRUE(caller_video_sender->SetTrack(caller()->CreateLocalVideoTrack()));
ASSERT_TRUE(callee_audio_sender->SetTrack(callee()->CreateLocalAudioTrack()));
ASSERT_TRUE(callee_video_sender->SetTrack(callee()->CreateLocalVideoTrack()));
ASSERT_TRUE(
caller_audio_sender->SetTrack(caller()->CreateLocalAudioTrack().get()));
ASSERT_TRUE(
caller_video_sender->SetTrack(caller()->CreateLocalVideoTrack().get()));
ASSERT_TRUE(
callee_audio_sender->SetTrack(callee()->CreateLocalAudioTrack().get()));
ASSERT_TRUE(
callee_video_sender->SetTrack(callee()->CreateLocalVideoTrack().get()));
MediaExpectations media_expectations;
media_expectations.ExpectBidirectionalAudioAndVideo();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
@ -2841,7 +2850,7 @@ TEST_P(PeerConnectionIntegrationTest, IceTransportFactoryUsedForConnections) {
ASSERT_TRUE(wrapper);
wrapper->CreateDataChannel();
auto observer = rtc::make_ref_counted<MockSetSessionDescriptionObserver>();
wrapper->pc()->SetLocalDescription(observer,
wrapper->pc()->SetLocalDescription(observer.get(),
wrapper->CreateOfferAndWait().release());
}
@ -3326,7 +3335,7 @@ TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
caller()->AddVideoTrack();
callee()->AddVideoTrack();
auto observer = rtc::make_ref_counted<MockSetSessionDescriptionObserver>();
callee()->pc()->SetLocalDescription(observer,
callee()->pc()->SetLocalDescription(observer.get(),
callee()->CreateOfferAndWait().release());
EXPECT_TRUE_WAIT(observer->called(), kDefaultTimeout);
caller()->CreateAndSetAndSignalOffer(); // Implicit rollback.
@ -3344,7 +3353,7 @@ TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
auto sld_observer =
rtc::make_ref_counted<MockSetSessionDescriptionObserver>();
callee()->pc()->SetLocalDescription(sld_observer,
callee()->pc()->SetLocalDescription(sld_observer.get(),
callee()->CreateOfferAndWait().release());
EXPECT_TRUE_WAIT(sld_observer->called(), kDefaultTimeout);
EXPECT_EQ(sld_observer->error(), "");
@ -3352,7 +3361,7 @@ TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
auto srd_observer =
rtc::make_ref_counted<MockSetSessionDescriptionObserver>();
callee()->pc()->SetRemoteDescription(
srd_observer, caller()->CreateOfferAndWait().release());
srd_observer.get(), caller()->CreateOfferAndWait().release());
EXPECT_TRUE_WAIT(srd_observer->called(), kDefaultTimeout);
EXPECT_EQ(srd_observer->error(), "");

139
third_party/libwebrtc/pc/peer_connection_interface_unittest.cc поставляемый
Просмотреть файл

@ -549,14 +549,14 @@ rtc::scoped_refptr<StreamCollection> CreateStreamCollection(
rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
webrtc::AudioTrack::Create(kAudioTracks[i * tracks_per_stream + j],
nullptr));
stream->AddTrack(audio_track);
stream->AddTrack(audio_track.get());
// Add a local video track.
rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track(
webrtc::VideoTrack::Create(kVideoTracks[i * tracks_per_stream + j],
webrtc::FakeVideoTrackSource::Create(),
rtc::Thread::Current()));
stream->AddTrack(video_track);
stream->AddTrack(video_track.get());
}
local_collection->AddStream(stream);
@ -810,13 +810,14 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test {
}
void ReleasePeerConnection() {
pc_ = NULL;
observer_.SetPeerConnectionInterface(NULL);
pc_ = nullptr;
observer_.SetPeerConnectionInterface(nullptr);
}
rtc::scoped_refptr<VideoTrackInterface> CreateVideoTrack(
const std::string& label) {
return pc_factory_->CreateVideoTrack(label, FakeVideoTrackSource::Create());
return pc_factory_->CreateVideoTrack(label,
FakeVideoTrackSource::Create().get());
}
void AddVideoTrack(const std::string& track_label,
@ -829,8 +830,8 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test {
void AddVideoStream(const std::string& label) {
rtc::scoped_refptr<MediaStreamInterface> stream(
pc_factory_->CreateLocalMediaStream(label));
stream->AddTrack(CreateVideoTrack(label + "v0"));
ASSERT_TRUE(pc_->AddStream(stream));
stream->AddTrack(CreateVideoTrack(label + "v0").get());
ASSERT_TRUE(pc_->AddStream(stream.get()));
}
rtc::scoped_refptr<AudioTrackInterface> CreateAudioTrack(
@ -848,8 +849,8 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test {
void AddAudioStream(const std::string& label) {
rtc::scoped_refptr<MediaStreamInterface> stream(
pc_factory_->CreateLocalMediaStream(label));
stream->AddTrack(CreateAudioTrack(label + "a0"));
ASSERT_TRUE(pc_->AddStream(stream));
stream->AddTrack(CreateAudioTrack(label + "a0").get());
ASSERT_TRUE(pc_->AddStream(stream.get()));
}
void AddAudioVideoStream(const std::string& stream_id,
@ -858,9 +859,9 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test {
// Create a local stream.
rtc::scoped_refptr<MediaStreamInterface> stream(
pc_factory_->CreateLocalMediaStream(stream_id));
stream->AddTrack(CreateAudioTrack(audio_track_label));
stream->AddTrack(CreateVideoTrack(video_track_label));
ASSERT_TRUE(pc_->AddStream(stream));
stream->AddTrack(CreateAudioTrack(audio_track_label).get());
stream->AddTrack(CreateVideoTrack(video_track_label).get());
ASSERT_TRUE(pc_->AddStream(stream.get()));
}
rtc::scoped_refptr<RtpReceiverInterface> GetFirstReceiverOfType(
@ -879,9 +880,11 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test {
auto observer =
rtc::make_ref_counted<MockCreateSessionDescriptionObserver>();
if (offer) {
pc_->CreateOffer(observer, options ? *options : RTCOfferAnswerOptions());
pc_->CreateOffer(observer.get(),
options ? *options : RTCOfferAnswerOptions());
} else {
pc_->CreateAnswer(observer, options ? *options : RTCOfferAnswerOptions());
pc_->CreateAnswer(observer.get(),
options ? *options : RTCOfferAnswerOptions());
}
EXPECT_EQ_WAIT(true, observer->called(), kTimeout);
*desc = observer->MoveDescription();
@ -903,9 +906,9 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test {
bool local) {
auto observer = rtc::make_ref_counted<MockSetSessionDescriptionObserver>();
if (local) {
pc_->SetLocalDescription(observer, desc.release());
pc_->SetLocalDescription(observer.get(), desc.release());
} else {
pc_->SetRemoteDescription(observer, desc.release());
pc_->SetRemoteDescription(observer.get(), desc.release());
}
if (pc_->signaling_state() != PeerConnectionInterface::kClosed) {
EXPECT_EQ_WAIT(true, observer->called(), kTimeout);
@ -928,7 +931,7 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test {
// be required.
bool DoGetStats(MediaStreamTrackInterface* track) {
auto observer = rtc::make_ref_counted<MockStatsObserver>();
if (!pc_->GetStats(observer, track,
if (!pc_->GetStats(observer.get(), track,
PeerConnectionInterface::kStatsOutputLevelStandard))
return false;
EXPECT_TRUE_WAIT(observer->called(), kTimeout);
@ -939,7 +942,7 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test {
bool DoGetRTCStats() {
auto callback =
rtc::make_ref_counted<webrtc::MockRTCStatsCollectorCallback>();
pc_->GetStats(callback);
pc_->GetStats(callback.get());
EXPECT_TRUE_WAIT(callback->called(), kTimeout);
return callback->called();
}
@ -963,12 +966,12 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test {
const cricket::MediaContentDescription* desc =
cricket::GetFirstAudioContentDescription(
pc_->remote_description()->description());
ASSERT_TRUE(desc != NULL);
ASSERT_TRUE(desc != nullptr);
EXPECT_GT(desc->rtp_header_extensions().size(), 0u);
desc = cricket::GetFirstVideoContentDescription(
pc_->remote_description()->description());
ASSERT_TRUE(desc != NULL);
ASSERT_TRUE(desc != nullptr);
EXPECT_GT(desc->rtp_header_extensions().size(), 0u);
}
@ -1115,21 +1118,21 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test {
if (number_of_audio_tracks > 0) {
sdp_ms1 += std::string(kSdpStringAudio);
sdp_ms1 += std::string(kSdpStringMs1Audio0);
AddAudioTrack(kAudioTracks[0], stream);
AddAudioTrack(kAudioTracks[0], stream.get());
}
if (number_of_audio_tracks > 1) {
sdp_ms1 += kSdpStringMs1Audio1;
AddAudioTrack(kAudioTracks[1], stream);
AddAudioTrack(kAudioTracks[1], stream.get());
}
if (number_of_video_tracks > 0) {
sdp_ms1 += std::string(kSdpStringVideo);
sdp_ms1 += std::string(kSdpStringMs1Video0);
AddVideoTrack(kVideoTracks[0], stream);
AddVideoTrack(kVideoTracks[0], stream.get());
}
if (number_of_video_tracks > 1) {
sdp_ms1 += kSdpStringMs1Video1;
AddVideoTrack(kVideoTracks[1], stream);
AddVideoTrack(kVideoTracks[1], stream.get());
}
return std::unique_ptr<SessionDescriptionInterface>(
@ -1140,7 +1143,7 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test {
MediaStreamInterface* stream) {
rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
webrtc::AudioTrack::Create(track_id, nullptr));
ASSERT_TRUE(stream->AddTrack(audio_track));
ASSERT_TRUE(stream->AddTrack(audio_track.get()));
}
void AddVideoTrack(const std::string& track_id,
@ -1149,7 +1152,7 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test {
webrtc::VideoTrack::Create(track_id,
webrtc::FakeVideoTrackSource::Create(),
rtc::Thread::Current()));
ASSERT_TRUE(stream->AddTrack(video_track));
ASSERT_TRUE(stream->AddTrack(video_track.get()));
}
std::unique_ptr<SessionDescriptionInterface> CreateOfferWithOneAudioTrack() {
@ -1195,7 +1198,7 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test {
RTC_DCHECK(pc_);
auto observer =
rtc::make_ref_counted<MockCreateSessionDescriptionObserver>();
pc_->CreateOffer(observer, offer_answer_options);
pc_->CreateOffer(observer.get(), offer_answer_options);
EXPECT_EQ_WAIT(true, observer->called(), kTimeout);
return observer->MoveDescription();
}
@ -1394,7 +1397,7 @@ TEST_P(PeerConnectionInterfaceTest,
auto result = pc_factory_->CreatePeerConnectionOrError(
config, std::move(pc_dependencies));
EXPECT_TRUE(result.ok());
observer_.SetPeerConnectionInterface(result.value());
observer_.SetPeerConnectionInterface(result.value().get());
// Now validate that the config fields set above were applied to the
// PortAllocator, as flags or otherwise.
@ -1458,10 +1461,10 @@ TEST_F(PeerConnectionInterfaceTestPlanB, AddStreams) {
rtc::scoped_refptr<MediaStreamInterface> stream(
pc_factory_->CreateLocalMediaStream(kStreamId3));
rtc::scoped_refptr<AudioTrackInterface> audio_track(
pc_factory_->CreateAudioTrack(kStreamId3,
static_cast<AudioSourceInterface*>(NULL)));
pc_factory_->CreateAudioTrack(
kStreamId3, static_cast<AudioSourceInterface*>(nullptr)));
stream->AddTrack(audio_track.get());
EXPECT_TRUE(pc_->AddStream(stream));
EXPECT_TRUE(pc_->AddStream(stream.get()));
EXPECT_EQ(3u, pc_->local_streams()->count());
// Remove the third stream.
@ -1824,7 +1827,7 @@ TEST_F(PeerConnectionInterfaceTestPlanB, RemoveTrackAfterAddStream) {
MediaStreamInterface* stream = pc_->local_streams()->at(0);
// Remove the video track.
stream->RemoveTrack(stream->GetVideoTracks()[0]);
stream->RemoveTrack(stream->GetVideoTracks()[0].get());
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
@ -1858,7 +1861,7 @@ TEST_P(PeerConnectionInterfaceTest, GetStatsForSpecificTrack) {
ASSERT_LT(0u, pc_->GetReceivers().size());
rtc::scoped_refptr<MediaStreamTrackInterface> remote_audio =
pc_->GetReceivers()[0]->track();
EXPECT_TRUE(DoGetStats(remote_audio));
EXPECT_TRUE(DoGetStats(remote_audio.get()));
// Remove the stream. Since we are sending to our selves the local
// and the remote stream is the same.
@ -1868,7 +1871,7 @@ TEST_P(PeerConnectionInterfaceTest, GetStatsForSpecificTrack) {
// Test that we still can get statistics for the old track. Even if it is not
// sent any longer.
EXPECT_TRUE(DoGetStats(remote_audio));
EXPECT_TRUE(DoGetStats(remote_audio.get()));
}
// Test that we can get stats on a video track.
@ -1876,15 +1879,15 @@ TEST_P(PeerConnectionInterfaceTest, GetStatsForVideoTrack) {
InitiateCall();
auto video_receiver = GetFirstReceiverOfType(cricket::MEDIA_TYPE_VIDEO);
ASSERT_TRUE(video_receiver);
EXPECT_TRUE(DoGetStats(video_receiver->track()));
EXPECT_TRUE(DoGetStats(video_receiver->track().get()));
}
// Test that we don't get statistics for an invalid track.
TEST_P(PeerConnectionInterfaceTest, GetStatsForInvalidTrack) {
InitiateCall();
rtc::scoped_refptr<AudioTrackInterface> unknown_audio_track(
pc_factory_->CreateAudioTrack("unknown track", NULL));
EXPECT_FALSE(DoGetStats(unknown_audio_track));
pc_factory_->CreateAudioTrack("unknown track", nullptr));
EXPECT_FALSE(DoGetStats(unknown_audio_track.get()));
}
TEST_P(PeerConnectionInterfaceTest, GetRTCStatsBeforeAndAfterCalling) {
@ -2048,22 +2051,22 @@ TEST_P(PeerConnectionInterfaceTest, ReceiveFireFoxOffer) {
webrtc::kFireFoxSdpOffer, nullptr));
EXPECT_TRUE(DoSetSessionDescription(std::move(desc), false));
CreateAnswerAsLocalDescription();
ASSERT_TRUE(pc_->local_description() != NULL);
ASSERT_TRUE(pc_->remote_description() != NULL);
ASSERT_TRUE(pc_->local_description() != nullptr);
ASSERT_TRUE(pc_->remote_description() != nullptr);
const cricket::ContentInfo* content =
cricket::GetFirstAudioContent(pc_->local_description()->description());
ASSERT_TRUE(content != NULL);
ASSERT_TRUE(content != nullptr);
EXPECT_FALSE(content->rejected);
content =
cricket::GetFirstVideoContent(pc_->local_description()->description());
ASSERT_TRUE(content != NULL);
ASSERT_TRUE(content != nullptr);
EXPECT_FALSE(content->rejected);
#ifdef WEBRTC_HAVE_SCTP
content =
cricket::GetFirstDataContent(pc_->local_description()->description());
ASSERT_TRUE(content != NULL);
ASSERT_TRUE(content != nullptr);
EXPECT_FALSE(content->rejected);
#endif
}
@ -2452,13 +2455,13 @@ TEST_F(PeerConnectionInterfaceTestPlanB, CloseAndTestMethods) {
pc_->Close();
pc_->RemoveStream(local_stream);
EXPECT_FALSE(pc_->AddStream(local_stream));
pc_->RemoveStream(local_stream.get());
EXPECT_FALSE(pc_->AddStream(local_stream.get()));
EXPECT_FALSE(pc_->CreateDataChannelOrError("test", NULL).ok());
EXPECT_TRUE(pc_->local_description() != NULL);
EXPECT_TRUE(pc_->remote_description() != NULL);
EXPECT_TRUE(pc_->local_description() != nullptr);
EXPECT_TRUE(pc_->remote_description() != nullptr);
std::unique_ptr<SessionDescriptionInterface> offer;
EXPECT_FALSE(DoCreateOffer(&offer, nullptr));
@ -2481,7 +2484,7 @@ TEST_F(PeerConnectionInterfaceTestPlanB, CloseAndTestMethods) {
TEST_P(PeerConnectionInterfaceTest, CloseAndGetStats) {
InitiateCall();
pc_->Close();
DoGetStats(NULL);
DoGetStats(nullptr);
}
// NOTE: The series of tests below come from what used to be
@ -2523,14 +2526,14 @@ TEST_F(PeerConnectionInterfaceTestPlanB,
CreateSessionDescriptionAndReference(1, 1);
EXPECT_TRUE(DoSetRemoteDescription(std::move(desc_ms1)));
EXPECT_TRUE(CompareStreamCollections(observer_.remote_streams(),
reference_collection_));
reference_collection_.get()));
// Add extra audio and video tracks to the same MediaStream.
std::unique_ptr<SessionDescriptionInterface> desc_ms1_two_tracks =
CreateSessionDescriptionAndReference(2, 2);
EXPECT_TRUE(DoSetRemoteDescription(std::move(desc_ms1_two_tracks)));
EXPECT_TRUE(CompareStreamCollections(observer_.remote_streams(),
reference_collection_));
reference_collection_.get()));
rtc::scoped_refptr<AudioTrackInterface> audio_track2 =
observer_.remote_streams()->at(0)->GetAudioTracks()[1];
EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, audio_track2->state());
@ -2541,14 +2544,14 @@ TEST_F(PeerConnectionInterfaceTestPlanB,
// Remove the extra audio and video tracks.
std::unique_ptr<SessionDescriptionInterface> desc_ms2 =
CreateSessionDescriptionAndReference(1, 1);
MockTrackObserver audio_track_observer(audio_track2);
MockTrackObserver video_track_observer(video_track2);
MockTrackObserver audio_track_observer(audio_track2.get());
MockTrackObserver video_track_observer(video_track2.get());
EXPECT_CALL(audio_track_observer, OnChanged()).Times(Exactly(1));
EXPECT_CALL(video_track_observer, OnChanged()).Times(Exactly(1));
EXPECT_TRUE(DoSetRemoteDescription(std::move(desc_ms2)));
EXPECT_TRUE(CompareStreamCollections(observer_.remote_streams(),
reference_collection_));
reference_collection_.get()));
// Track state may be updated asynchronously.
EXPECT_EQ_WAIT(webrtc::MediaStreamTrackInterface::kEnded,
audio_track2->state(), kTimeout);
@ -2611,8 +2614,8 @@ TEST_F(PeerConnectionInterfaceTestPlanB, RemoveTrackThenRejectMediaContent) {
CreatePeerConnection(config);
CreateAndSetRemoteOffer(GetSdpStringWithStream1());
MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
remote_stream->RemoveTrack(remote_stream->GetVideoTracks()[0]);
remote_stream->RemoveTrack(remote_stream->GetAudioTracks()[0]);
remote_stream->RemoveTrack(remote_stream->GetVideoTracks()[0].get());
remote_stream->RemoveTrack(remote_stream->GetAudioTracks()[0].get());
std::unique_ptr<SessionDescriptionInterface> local_answer(
webrtc::CreateSessionDescription(SdpType::kAnswer,
@ -2696,8 +2699,8 @@ TEST_F(PeerConnectionInterfaceTestPlanB, RemoveAlreadyGoneRemoteStream) {
CreatePeerConnection(config);
CreateAndSetRemoteOffer(GetSdpStringWithStream1());
MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
remote_stream->RemoveTrack(remote_stream->GetAudioTracks()[0]);
remote_stream->RemoveTrack(remote_stream->GetVideoTracks()[0]);
remote_stream->RemoveTrack(remote_stream->GetAudioTracks()[0].get());
remote_stream->RemoveTrack(remote_stream->GetVideoTracks()[0].get());
CreateAndSetRemoteOffer(kSdpStringWithoutStreams);
@ -2918,7 +2921,7 @@ TEST_P(PeerConnectionInterfaceTest,
// Change the ssrc of the audio and video track.
cricket::MediaContentDescription* desc =
cricket::GetFirstAudioContentDescription(modified_offer->description());
ASSERT_TRUE(desc != NULL);
ASSERT_TRUE(desc != nullptr);
for (StreamParams& stream : desc->mutable_streams()) {
for (unsigned int& ssrc : stream.ssrcs) {
++ssrc;
@ -2927,7 +2930,7 @@ TEST_P(PeerConnectionInterfaceTest,
desc =
cricket::GetFirstVideoContentDescription(modified_offer->description());
ASSERT_TRUE(desc != NULL);
ASSERT_TRUE(desc != nullptr);
for (StreamParams& stream : desc->mutable_streams()) {
for (unsigned int& ssrc : stream.ssrcs) {
++ssrc;
@ -2966,9 +2969,9 @@ TEST_F(PeerConnectionInterfaceTestPlanB,
// Add a new MediaStream but with the same tracks as in the first stream.
rtc::scoped_refptr<webrtc::MediaStreamInterface> stream_1(
webrtc::MediaStream::Create(kStreams[1]));
stream_1->AddTrack(stream_collection->at(0)->GetVideoTracks()[0]);
stream_1->AddTrack(stream_collection->at(0)->GetAudioTracks()[0]);
pc_->AddStream(stream_1);
stream_1->AddTrack(stream_collection->at(0)->GetVideoTracks()[0].get());
stream_1->AddTrack(stream_collection->at(0)->GetAudioTracks()[0].get());
pc_->AddStream(stream_1.get());
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
EXPECT_TRUE(DoSetLocalDescription(std::move(offer)));
@ -3576,27 +3579,27 @@ TEST_F(PeerConnectionInterfaceTestPlanB,
CreatePeerConnectionWithoutDtls();
rtc::scoped_refptr<MediaStreamInterface> stream(
pc_factory_->CreateLocalMediaStream(kStreamId1));
pc_->AddStream(stream);
pc_->AddStream(stream.get());
rtc::scoped_refptr<AudioTrackInterface> audio_track(
CreateAudioTrack("audio_track"));
rtc::scoped_refptr<VideoTrackInterface> video_track(
CreateVideoTrack("video_track"));
stream->AddTrack(audio_track);
stream->AddTrack(audio_track.get());
EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout);
observer_.renegotiation_needed_ = false;
CreateOfferReceiveAnswer();
stream->AddTrack(video_track);
stream->AddTrack(video_track.get());
EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout);
observer_.renegotiation_needed_ = false;
CreateOfferReceiveAnswer();
stream->RemoveTrack(audio_track);
stream->RemoveTrack(audio_track.get());
EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout);
observer_.renegotiation_needed_ = false;
CreateOfferReceiveAnswer();
stream->RemoveTrack(video_track);
stream->RemoveTrack(video_track.get());
EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout);
observer_.renegotiation_needed_ = false;
}
@ -3663,7 +3666,7 @@ class PeerConnectionMediaConfigTest : public ::testing::Test {
auto result =
pcf_->CreatePeerConnectionOrError(config, std::move(pc_dependencies));
EXPECT_TRUE(result.ok());
observer_.SetPeerConnectionInterface(result.value());
observer_.SetPeerConnectionInterface(result.value().get());
return result.value()->GetConfiguration().media_config;
}
@ -3679,7 +3682,7 @@ TEST_F(PeerConnectionMediaConfigTest, TestCreateAndClose) {
auto result =
pcf_->CreatePeerConnectionOrError(config, std::move(pc_dependencies));
EXPECT_TRUE(result.ok());
observer_.SetPeerConnectionInterface(result.value());
observer_.SetPeerConnectionInterface(result.value().get());
result.value()->Close(); // No abort -> ok.
SUCCEED();
}

4
third_party/libwebrtc/pc/peer_connection_jsep_unittest.cc поставляемый
Просмотреть файл

@ -123,7 +123,7 @@ class PeerConnectionJsepTest : public ::testing::Test {
return nullptr;
}
observer->SetPeerConnectionInterface(result.value());
observer->SetPeerConnectionInterface(result.value().get());
return std::make_unique<PeerConnectionWrapper>(
pc_factory, result.MoveValue(), std::move(observer));
}
@ -1363,7 +1363,7 @@ TEST_F(PeerConnectionJsepTest,
auto caller = CreatePeerConnection();
auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
transceiver->sender()->SetTrack(caller->CreateAudioTrack(kTrackId));
transceiver->sender()->SetTrack(caller->CreateAudioTrack(kTrackId).get());
auto offer = caller->CreateOffer();
auto contents = offer->description()->contents();

2
third_party/libwebrtc/pc/peer_connection_message_handler.cc поставляемый
Просмотреть файл

@ -117,7 +117,7 @@ void PeerConnectionMessageHandler::OnMessage(rtc::Message* msg) {
case MSG_GETSTATS: {
GetStatsMsg* param = static_cast<GetStatsMsg*>(msg->pdata);
StatsReports reports;
param->stats->GetStats(param->track, &reports);
param->stats->GetStats(param->track.get(), &reports);
param->observer->OnComplete(reports);
delete param;
break;

5
third_party/libwebrtc/pc/peer_connection_rampup_tests.cc поставляемый
Просмотреть файл

@ -123,14 +123,15 @@ class PeerConnectionWrapperForRampUpTest : public PeerConnectionWrapper {
video_track_sources_.back()->Start();
return rtc::scoped_refptr<VideoTrackInterface>(
pc_factory()->CreateVideoTrack(rtc::CreateRandomUuid(),
video_track_sources_.back()));
video_track_sources_.back().get()));
}
rtc::scoped_refptr<AudioTrackInterface> CreateLocalAudioTrack(
const cricket::AudioOptions options) {
rtc::scoped_refptr<AudioSourceInterface> source =
pc_factory()->CreateAudioSource(options);
return pc_factory()->CreateAudioTrack(rtc::CreateRandomUuid(), source);
return pc_factory()->CreateAudioTrack(rtc::CreateRandomUuid(),
source.get());
}
private:

6
third_party/libwebrtc/pc/peer_connection_rtp_unittest.cc поставляемый
Просмотреть файл

@ -138,7 +138,7 @@ class PeerConnectionRtpBaseTest : public ::testing::Test {
auto result = pc_factory_->CreatePeerConnectionOrError(
config, PeerConnectionDependencies(observer.get()));
EXPECT_TRUE(result.ok());
observer->SetPeerConnectionInterface(result.value());
observer->SetPeerConnectionInterface(result.value().get());
return std::make_unique<PeerConnectionWrapper>(
pc_factory_, result.MoveValue(), std::move(observer));
}
@ -921,7 +921,7 @@ TEST_P(PeerConnectionRtpTest,
rtc::make_ref_counted<webrtc::MockSetSessionDescriptionObserver>();
auto offer = caller->CreateOfferAndSetAsLocal();
callee->pc()->SetRemoteDescription(observer, offer.release());
callee->pc()->SetRemoteDescription(observer.get(), offer.release());
callee = nullptr;
rtc::Thread::Current()->ProcessMessages(0);
EXPECT_FALSE(observer->called());
@ -1989,7 +1989,7 @@ TEST_P(PeerConnectionRtpTest, CreateTwoSendersWithSameTrack) {
EXPECT_TRUE(sender1->SetTrack(nullptr));
auto sender2 = caller->AddTrack(track);
EXPECT_TRUE(sender2);
EXPECT_TRUE(sender1->SetTrack(track));
EXPECT_TRUE(sender1->SetTrack(track.get()));
if (sdp_semantics_ == SdpSemantics::kPlanB_DEPRECATED) {
// TODO(hbos): When https://crbug.com/webrtc/8734 is resolved, this should

23
third_party/libwebrtc/pc/peer_connection_signaling_unittest.cc поставляемый
Просмотреть файл

@ -153,7 +153,7 @@ class PeerConnectionSignalingBaseTest : public ::testing::Test {
return nullptr;
}
observer->SetPeerConnectionInterface(result.value());
observer->SetPeerConnectionInterface(result.value().get());
return std::make_unique<PeerConnectionWrapperForSignalingTest>(
pc_factory_, result.MoveValue(), std::move(observer));
}
@ -571,7 +571,7 @@ TEST_P(PeerConnectionSignalingTest, CreateOffersAndShutdown) {
rtc::scoped_refptr<MockCreateSessionDescriptionObserver> observers[100];
for (auto& observer : observers) {
observer = rtc::make_ref_counted<MockCreateSessionDescriptionObserver>();
caller->pc()->CreateOffer(observer, options);
caller->pc()->CreateOffer(observer.get(), options);
}
// Destroy the PeerConnection.
@ -593,7 +593,7 @@ TEST_P(PeerConnectionSignalingTest, CloseCreateOfferAndShutdown) {
auto caller = CreatePeerConnection();
auto observer = rtc::make_ref_counted<MockCreateSessionDescriptionObserver>();
caller->pc()->Close();
caller->pc()->CreateOffer(observer, RTCOfferAnswerOptions());
caller->pc()->CreateOffer(observer.get(), RTCOfferAnswerOptions());
caller.reset(nullptr);
EXPECT_TRUE(observer->called());
}
@ -680,7 +680,7 @@ TEST_P(PeerConnectionSignalingTest,
auto offer = caller->CreateOffer(RTCOfferAnswerOptions());
auto observer = MockSetSessionDescriptionObserver::Create();
caller->pc()->SetLocalDescription(observer, offer.release());
caller->pc()->SetLocalDescription(observer.get(), offer.release());
// The old observer is not invoked immediately.
EXPECT_FALSE(observer->called());
// Process all currently pending messages by waiting for a posted task to run.
@ -720,7 +720,7 @@ TEST_P(PeerConnectionSignalingTest, CreateOfferBlocksSetRemoteDescription) {
// Synchronously invoke CreateOffer() and SetRemoteDescription(). The
// SetRemoteDescription() operation should be chained to be executed
// asynchronously, when CreateOffer() completes.
callee->pc()->CreateOffer(offer_observer, RTCOfferAnswerOptions());
callee->pc()->CreateOffer(offer_observer.get(), RTCOfferAnswerOptions());
callee->pc()->SetRemoteDescription(
std::move(offer),
rtc::make_ref_counted<FakeSetRemoteDescriptionObserver>());
@ -802,7 +802,7 @@ TEST_P(PeerConnectionSignalingTest,
auto callee_set_remote_description_observer =
MockSetSessionDescriptionObserver::Create();
callee->pc()->SetRemoteDescription(
callee_set_remote_description_observer,
callee_set_remote_description_observer.get(),
CloneSessionDescription(caller->pc()->pending_local_description())
.release());
@ -822,7 +822,7 @@ TEST_P(PeerConnectionSignalingTest,
auto caller_set_remote_description_observer =
MockSetSessionDescriptionObserver::Create();
caller->pc()->SetRemoteDescription(
caller_set_remote_description_observer,
caller_set_remote_description_observer.get(),
CloneSessionDescription(callee->pc()->current_local_description())
.release());
EXPECT_TRUE_WAIT(caller_set_remote_description_observer->called(),
@ -1126,7 +1126,7 @@ TEST_F(PeerConnectionSignalingUnifiedPlanTest,
// waiting for it would not ensure synchronicity.
RTC_DCHECK(!caller->pc()->GetTransceivers()[0]->mid().has_value());
caller->pc()->SetLocalDescription(
rtc::make_ref_counted<MockSetSessionDescriptionObserver>(),
rtc::make_ref_counted<MockSetSessionDescriptionObserver>().get(),
offer.release());
EXPECT_TRUE(caller->pc()->GetTransceivers()[0]->mid().has_value());
}
@ -1162,11 +1162,12 @@ TEST_F(PeerConnectionSignalingUnifiedPlanTest,
// operation executed immediately.
RTC_DCHECK(!pc->GetTransceivers()[0]->mid().has_value());
pc->SetLocalDescription(
rtc::make_ref_counted<MockSetSessionDescriptionObserver>(),
rtc::make_ref_counted<MockSetSessionDescriptionObserver>()
.get(),
desc);
EXPECT_TRUE(pc->GetTransceivers()[0]->mid().has_value());
});
caller->pc()->CreateOffer(offer_observer, RTCOfferAnswerOptions());
caller->pc()->CreateOffer(offer_observer.get(), RTCOfferAnswerOptions());
EXPECT_TRUE_WAIT(offer_observer->was_called(), kWaitTimeout);
}
@ -1253,7 +1254,7 @@ TEST_F(PeerConnectionSignalingUnifiedPlanTest,
EXPECT_TRUE(caller->observer()->has_negotiation_needed_event());
auto observer = rtc::make_ref_counted<MockCreateSessionDescriptionObserver>();
caller->pc()->CreateOffer(observer, RTCOfferAnswerOptions());
caller->pc()->CreateOffer(observer.get(), RTCOfferAnswerOptions());
// For this test to work, the operation has to be pending, i.e. the observer
// has not yet been invoked.
EXPECT_FALSE(observer->called());

2
third_party/libwebrtc/pc/peer_connection_simulcast_unittest.cc поставляемый
Просмотреть файл

@ -140,7 +140,7 @@ class PeerConnectionSimulcastTests : public ::testing::Test {
auto result =
pc_factory_->CreatePeerConnectionOrError(config, std::move(pcd));
EXPECT_TRUE(result.ok());
observer->SetPeerConnectionInterface(result.value());
observer->SetPeerConnectionInterface(result.value().get());
return result.MoveValue();
}

9
third_party/libwebrtc/pc/peer_connection_wrapper.cc поставляемый
Просмотреть файл

@ -135,7 +135,7 @@ std::unique_ptr<SessionDescriptionInterface> PeerConnectionWrapper::CreateSdp(
rtc::FunctionView<void(CreateSessionDescriptionObserver*)> fn,
std::string* error_out) {
auto observer = rtc::make_ref_counted<MockCreateSessionDescriptionObserver>();
fn(observer);
fn(observer.get());
EXPECT_EQ_WAIT(true, observer->called(), kDefaultTimeout);
if (error_out && !observer->result()) {
*error_out = observer->error();
@ -179,7 +179,7 @@ bool PeerConnectionWrapper::SetSdp(
rtc::FunctionView<void(SetSessionDescriptionObserver*)> fn,
std::string* error_out) {
auto observer = rtc::make_ref_counted<MockSetSessionDescriptionObserver>();
fn(observer);
fn(observer.get());
EXPECT_EQ_WAIT(true, observer->called(), kDefaultTimeout);
if (error_out && !observer->result()) {
*error_out = observer->error();
@ -277,7 +277,8 @@ rtc::scoped_refptr<AudioTrackInterface> PeerConnectionWrapper::CreateAudioTrack(
rtc::scoped_refptr<VideoTrackInterface> PeerConnectionWrapper::CreateVideoTrack(
const std::string& label) {
return pc_factory()->CreateVideoTrack(label, FakeVideoTrackSource::Create());
return pc_factory()->CreateVideoTrack(label,
FakeVideoTrackSource::Create().get());
}
rtc::scoped_refptr<RtpSenderInterface> PeerConnectionWrapper::AddTrack(
@ -329,7 +330,7 @@ bool PeerConnectionWrapper::IsIceConnected() {
rtc::scoped_refptr<const webrtc::RTCStatsReport>
PeerConnectionWrapper::GetStats() {
auto callback = rtc::make_ref_counted<MockRTCStatsCollectorCallback>();
pc()->GetStats(callback);
pc()->GetStats(callback.get());
EXPECT_TRUE_WAIT(callback->called(), kDefaultTimeout);
return callback->report();
}

12
third_party/libwebrtc/pc/rtc_stats_collector.cc поставляемый
Просмотреть файл

@ -1893,7 +1893,8 @@ void RTCStatsCollector::ProduceAudioRTPStreamStats_n(
inbound_audio->track_id =
RTCMediaStreamTrackStatsIDFromDirectionAndAttachment(
kReceiver,
track_media_info_map.GetAttachmentIdByTrack(audio_track).value());
track_media_info_map.GetAttachmentIdByTrack(audio_track.get())
.value());
}
inbound_audio->transport_id = transport_id;
// Remote-outbound.
@ -1924,7 +1925,8 @@ void RTCStatsCollector::ProduceAudioRTPStreamStats_n(
track_media_info_map.GetAudioTrack(voice_sender_info);
if (audio_track) {
int attachment_id =
track_media_info_map.GetAttachmentIdByTrack(audio_track).value();
track_media_info_map.GetAttachmentIdByTrack(audio_track.get())
.value();
outbound_audio->track_id =
RTCMediaStreamTrackStatsIDFromDirectionAndAttachment(kSender,
attachment_id);
@ -1985,7 +1987,8 @@ void RTCStatsCollector::ProduceVideoRTPStreamStats_n(
inbound_video->track_id =
RTCMediaStreamTrackStatsIDFromDirectionAndAttachment(
kReceiver,
track_media_info_map.GetAttachmentIdByTrack(video_track).value());
track_media_info_map.GetAttachmentIdByTrack(video_track.get())
.value());
}
inbound_video->transport_id = transport_id;
report->AddStats(std::move(inbound_video));
@ -2007,7 +2010,8 @@ void RTCStatsCollector::ProduceVideoRTPStreamStats_n(
track_media_info_map.GetVideoTrack(video_sender_info);
if (video_track) {
int attachment_id =
track_media_info_map.GetAttachmentIdByTrack(video_track).value();
track_media_info_map.GetAttachmentIdByTrack(video_track.get())
.value();
outbound_video->track_id =
RTCMediaStreamTrackStatsIDFromDirectionAndAttachment(kSender,
attachment_id);

45
third_party/libwebrtc/pc/rtc_stats_collector_unittest.cc поставляемый
Просмотреть файл

@ -427,8 +427,8 @@ class RTCStatsCollectorWrapper {
rtc::scoped_refptr<FakePeerConnectionForStats> pc)
: pc_(pc),
stats_collector_(
RTCStatsCollector::Create(pc, 50 * rtc::kNumMicrosecsPerMillisec)) {
}
RTCStatsCollector::Create(pc.get(),
50 * rtc::kNumMicrosecsPerMillisec)) {}
rtc::scoped_refptr<RTCStatsCollector> stats_collector() {
return stats_collector_;
@ -631,7 +631,7 @@ class RTCStatsCollectorWrapper {
private:
rtc::scoped_refptr<const RTCStatsReport> WaitForReport(
rtc::scoped_refptr<RTCStatsObtainer> callback) {
EXPECT_TRUE_WAIT(callback->report(), kGetStatsReportTimeoutMs);
EXPECT_TRUE_WAIT(callback->report() != nullptr, kGetStatsReportTimeoutMs);
int64_t after = rtc::TimeUTCMicros();
for (const RTCStats& stats : *callback->report()) {
if (stats.type() == RTCRemoteInboundRtpStreamStats::kType ||
@ -920,7 +920,7 @@ class RTCStatsCollectorTest : public ::testing::Test {
TEST_F(RTCStatsCollectorTest, SingleCallback) {
rtc::scoped_refptr<const RTCStatsReport> result;
stats_->stats_collector()->GetStatsReport(RTCStatsObtainer::Create(&result));
EXPECT_TRUE_WAIT(result, kGetStatsReportTimeoutMs);
EXPECT_TRUE_WAIT(result != nullptr, kGetStatsReportTimeoutMs);
}
TEST_F(RTCStatsCollectorTest, MultipleCallbacks) {
@ -928,9 +928,9 @@ TEST_F(RTCStatsCollectorTest, MultipleCallbacks) {
stats_->stats_collector()->GetStatsReport(RTCStatsObtainer::Create(&a));
stats_->stats_collector()->GetStatsReport(RTCStatsObtainer::Create(&b));
stats_->stats_collector()->GetStatsReport(RTCStatsObtainer::Create(&c));
EXPECT_TRUE_WAIT(a, kGetStatsReportTimeoutMs);
EXPECT_TRUE_WAIT(b, kGetStatsReportTimeoutMs);
EXPECT_TRUE_WAIT(c, kGetStatsReportTimeoutMs);
EXPECT_TRUE_WAIT(a != nullptr, kGetStatsReportTimeoutMs);
EXPECT_TRUE_WAIT(b != nullptr, kGetStatsReportTimeoutMs);
EXPECT_TRUE_WAIT(c != nullptr, kGetStatsReportTimeoutMs);
EXPECT_EQ(a.get(), b.get());
EXPECT_EQ(b.get(), c.get());
@ -959,9 +959,9 @@ TEST_F(RTCStatsCollectorTest, MultipleCallbacksWithInvalidatedCacheInBetween) {
// Cache is invalidated after 50 ms.
fake_clock_.AdvanceTime(TimeDelta::Millis(51));
stats_->stats_collector()->GetStatsReport(RTCStatsObtainer::Create(&c));
EXPECT_TRUE_WAIT(a, kGetStatsReportTimeoutMs);
EXPECT_TRUE_WAIT(b, kGetStatsReportTimeoutMs);
EXPECT_TRUE_WAIT(c, kGetStatsReportTimeoutMs);
EXPECT_TRUE_WAIT(a != nullptr, kGetStatsReportTimeoutMs);
EXPECT_TRUE_WAIT(b != nullptr, kGetStatsReportTimeoutMs);
EXPECT_TRUE_WAIT(c != nullptr, kGetStatsReportTimeoutMs);
EXPECT_EQ(a.get(), b.get());
// The act of doing `AdvanceTime` processes all messages. If this was not the
// case we might not require `c` to be fresher than `b`.
@ -1748,10 +1748,10 @@ TEST_F(RTCStatsCollectorTest,
rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
RTCMediaStreamStats expected_local_stream(
IdForType<RTCMediaStreamStats>(report), report->timestamp_us());
IdForType<RTCMediaStreamStats>(report.get()), report->timestamp_us());
expected_local_stream.stream_identifier = local_stream->id();
expected_local_stream.track_ids = {
IdForType<RTCMediaStreamTrackStats>(report)};
IdForType<RTCMediaStreamTrackStats>(report.get())};
ASSERT_TRUE(report->Get(expected_local_stream.id()))
<< "Did not find " << expected_local_stream.id() << " in "
<< report->ToJson();
@ -1760,7 +1760,7 @@ TEST_F(RTCStatsCollectorTest,
report->Get(expected_local_stream.id())->cast_to<RTCMediaStreamStats>());
RTCMediaStreamTrackStats expected_local_audio_track_ssrc1(
IdForType<RTCMediaStreamTrackStats>(report), report->timestamp_us(),
IdForType<RTCMediaStreamTrackStats>(report.get()), report->timestamp_us(),
RTCMediaStreamTrackKind::kAudio);
expected_local_audio_track_ssrc1.track_identifier = local_audio_track->id();
expected_local_audio_track_ssrc1.media_source_id =
@ -1819,10 +1819,10 @@ TEST_F(RTCStatsCollectorTest,
rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
RTCMediaStreamStats expected_remote_stream(
IdForType<RTCMediaStreamStats>(report), report->timestamp_us());
IdForType<RTCMediaStreamStats>(report.get()), report->timestamp_us());
expected_remote_stream.stream_identifier = remote_stream->id();
expected_remote_stream.track_ids =
std::vector<std::string>({IdForType<RTCMediaStreamTrackStats>(report)});
expected_remote_stream.track_ids = std::vector<std::string>(
{IdForType<RTCMediaStreamTrackStats>(report.get())});
ASSERT_TRUE(report->Get(expected_remote_stream.id()))
<< "Did not find " << expected_remote_stream.id() << " in "
<< report->ToJson();
@ -1831,7 +1831,7 @@ TEST_F(RTCStatsCollectorTest,
report->Get(expected_remote_stream.id())->cast_to<RTCMediaStreamStats>());
RTCMediaStreamTrackStats expected_remote_audio_track(
IdForType<RTCMediaStreamTrackStats>(report), report->timestamp_us(),
IdForType<RTCMediaStreamTrackStats>(report.get()), report->timestamp_us(),
RTCMediaStreamTrackKind::kAudio);
expected_remote_audio_track.track_identifier = remote_audio_track->id();
// `expected_remote_audio_track.media_source_id` should be undefined
@ -2168,7 +2168,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Video) {
expected_video.ssrc = 1;
expected_video.media_type = "video";
expected_video.kind = "video";
expected_video.track_id = IdForType<RTCMediaStreamTrackStats>(report);
expected_video.track_id = IdForType<RTCMediaStreamTrackStats>(report.get());
expected_video.transport_id = "RTCTransport_TransportName_1";
expected_video.codec_id = "RTCCodec_VideoMid_Inbound_42";
expected_video.fir_count = 5;
@ -2259,7 +2259,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Audio) {
expected_audio.ssrc = 1;
expected_audio.media_type = "audio";
expected_audio.kind = "audio";
expected_audio.track_id = IdForType<RTCMediaStreamTrackStats>(report);
expected_audio.track_id = IdForType<RTCMediaStreamTrackStats>(report.get());
expected_audio.transport_id = "RTCTransport_TransportName_1";
expected_audio.codec_id = "RTCCodec_AudioMid_Outbound_42";
expected_audio.packets_sent = 2;
@ -2661,7 +2661,7 @@ TEST_F(RTCStatsCollectorTest, CollectNoStreamRTCOutboundRTPStreamStats_Audio) {
expected_audio.ssrc = 1;
expected_audio.media_type = "audio";
expected_audio.kind = "audio";
expected_audio.track_id = IdForType<RTCMediaStreamTrackStats>(report);
expected_audio.track_id = IdForType<RTCMediaStreamTrackStats>(report.get());
expected_audio.transport_id = "RTCTransport_TransportName_1";
expected_audio.codec_id = "RTCCodec_AudioMid_Outbound_42";
expected_audio.packets_sent = 2;
@ -3200,7 +3200,7 @@ TEST_F(RTCStatsCollectorTest, CollectEchoReturnLossFromTrackAudioProcessor) {
rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
RTCMediaStreamTrackStats expected_local_audio_track_ssrc1(
IdForType<RTCMediaStreamTrackStats>(report), report->timestamp_us(),
IdForType<RTCMediaStreamTrackStats>(report.get()), report->timestamp_us(),
RTCMediaStreamTrackKind::kAudio);
expected_local_audio_track_ssrc1.track_identifier = local_audio_track->id();
expected_local_audio_track_ssrc1.media_source_id =
@ -3493,7 +3493,8 @@ class FakeRTCStatsCollector : public RTCStatsCollector,
TEST(RTCStatsCollectorTestWithFakeCollector, ThreadUsageAndResultsMerging) {
auto pc = rtc::make_ref_counted<FakePeerConnectionForStats>();
rtc::scoped_refptr<FakeRTCStatsCollector> stats_collector(
FakeRTCStatsCollector::Create(pc, 50 * rtc::kNumMicrosecsPerMillisec));
FakeRTCStatsCollector::Create(pc.get(),
50 * rtc::kNumMicrosecsPerMillisec));
stats_collector->VerifyThreadUsageAndResultsMerging();
}

40
third_party/libwebrtc/pc/rtc_stats_integrationtest.cc поставляемый
Просмотреть файл

@ -174,8 +174,8 @@ class RTCStatsIntegrationTest : public ::testing::Test {
PeerConnectionInterface* pc) {
rtc::scoped_refptr<RTCStatsObtainer> stats_obtainer =
RTCStatsObtainer::Create();
pc->GetStats(stats_obtainer);
EXPECT_TRUE_WAIT(stats_obtainer->report(), kGetStatsTimeoutMs);
pc->GetStats(stats_obtainer.get());
EXPECT_TRUE_WAIT(stats_obtainer->report() != nullptr, kGetStatsTimeoutMs);
return stats_obtainer->report();
}
@ -186,7 +186,7 @@ class RTCStatsIntegrationTest : public ::testing::Test {
rtc::scoped_refptr<RTCStatsObtainer> stats_obtainer =
RTCStatsObtainer::Create();
pc->GetStats(selector, stats_obtainer);
EXPECT_TRUE_WAIT(stats_obtainer->report(), kGetStatsTimeoutMs);
EXPECT_TRUE_WAIT(stats_obtainer->report() != nullptr, kGetStatsTimeoutMs);
return stats_obtainer->report();
}
@ -432,7 +432,7 @@ class RTCStatsReportVerifier {
}
bool VerifyRTCCertificateStats(const RTCCertificateStats& certificate) {
RTCStatsVerifier verifier(report_, &certificate);
RTCStatsVerifier verifier(report_.get(), &certificate);
verifier.TestMemberIsDefined(certificate.fingerprint);
verifier.TestMemberIsDefined(certificate.fingerprint_algorithm);
verifier.TestMemberIsDefined(certificate.base64_certificate);
@ -442,7 +442,7 @@ class RTCStatsReportVerifier {
}
bool VerifyRTCCodecStats(const RTCCodecStats& codec) {
RTCStatsVerifier verifier(report_, &codec);
RTCStatsVerifier verifier(report_.get(), &codec);
verifier.TestMemberIsIDReference(codec.transport_id,
RTCTransportStats::kType);
verifier.TestMemberIsDefined(codec.payload_type);
@ -460,7 +460,7 @@ class RTCStatsReportVerifier {
}
bool VerifyRTCDataChannelStats(const RTCDataChannelStats& data_channel) {
RTCStatsVerifier verifier(report_, &data_channel);
RTCStatsVerifier verifier(report_.get(), &data_channel);
verifier.TestMemberIsDefined(data_channel.label);
verifier.TestMemberIsDefined(data_channel.protocol);
verifier.TestMemberIsDefined(data_channel.data_channel_identifier);
@ -475,7 +475,7 @@ class RTCStatsReportVerifier {
bool VerifyRTCIceCandidatePairStats(
const RTCIceCandidatePairStats& candidate_pair,
bool is_selected_pair) {
RTCStatsVerifier verifier(report_, &candidate_pair);
RTCStatsVerifier verifier(report_.get(), &candidate_pair);
verifier.TestMemberIsIDReference(candidate_pair.transport_id,
RTCTransportStats::kType);
verifier.TestMemberIsIDReference(candidate_pair.local_candidate_id,
@ -526,7 +526,7 @@ class RTCStatsReportVerifier {
}
bool VerifyRTCIceCandidateStats(const RTCIceCandidateStats& candidate) {
RTCStatsVerifier verifier(report_, &candidate);
RTCStatsVerifier verifier(report_.get(), &candidate);
verifier.TestMemberIsIDReference(candidate.transport_id,
RTCTransportStats::kType);
verifier.TestMemberIsDefined(candidate.is_remote);
@ -561,7 +561,7 @@ class RTCStatsReportVerifier {
}
bool VerifyRTCMediaStreamStats(const RTCMediaStreamStats& media_stream) {
RTCStatsVerifier verifier(report_, &media_stream);
RTCStatsVerifier verifier(report_.get(), &media_stream);
verifier.TestMemberIsDefined(media_stream.stream_identifier);
verifier.TestMemberIsIDReference(media_stream.track_ids,
RTCMediaStreamTrackStats::kType);
@ -570,7 +570,7 @@ class RTCStatsReportVerifier {
bool VerifyRTCMediaStreamTrackStats(
const RTCMediaStreamTrackStats& media_stream_track) {
RTCStatsVerifier verifier(report_, &media_stream_track);
RTCStatsVerifier verifier(report_.get(), &media_stream_track);
verifier.TestMemberIsDefined(media_stream_track.track_identifier);
verifier.TestMemberIsDefined(media_stream_track.remote_source);
verifier.TestMemberIsDefined(media_stream_track.ended);
@ -771,7 +771,7 @@ class RTCStatsReportVerifier {
bool VerifyRTCPeerConnectionStats(
const RTCPeerConnectionStats& peer_connection) {
RTCStatsVerifier verifier(report_, &peer_connection);
RTCStatsVerifier verifier(report_.get(), &peer_connection);
verifier.TestMemberIsNonNegative<uint32_t>(
peer_connection.data_channels_opened);
verifier.TestMemberIsNonNegative<uint32_t>(
@ -808,7 +808,7 @@ class RTCStatsReportVerifier {
bool VerifyRTCInboundRTPStreamStats(
const RTCInboundRTPStreamStats& inbound_stream) {
RTCStatsVerifier verifier(report_, &inbound_stream);
RTCStatsVerifier verifier(report_.get(), &inbound_stream);
VerifyRTCReceivedRtpStreamStats(inbound_stream, verifier,
inbound_stream.media_type.is_defined() &&
*inbound_stream.media_type == "audio");
@ -936,7 +936,7 @@ class RTCStatsReportVerifier {
bool VerifyRTCOutboundRTPStreamStats(
const RTCOutboundRTPStreamStats& outbound_stream) {
RTCStatsVerifier verifier(report_, &outbound_stream);
RTCStatsVerifier verifier(report_.get(), &outbound_stream);
VerifyRTCRTPStreamStats(outbound_stream, verifier);
if (outbound_stream.media_type.is_defined() &&
*outbound_stream.media_type == "video") {
@ -1048,7 +1048,7 @@ class RTCStatsReportVerifier {
bool VerifyRTCRemoteInboundRtpStreamStats(
const RTCRemoteInboundRtpStreamStats& remote_inbound_stream) {
RTCStatsVerifier verifier(report_, &remote_inbound_stream);
RTCStatsVerifier verifier(report_.get(), &remote_inbound_stream);
VerifyRTCReceivedRtpStreamStats(remote_inbound_stream, verifier, false);
verifier.TestMemberIsDefined(remote_inbound_stream.fraction_lost);
verifier.TestMemberIsIDReference(remote_inbound_stream.local_id,
@ -1064,7 +1064,7 @@ class RTCStatsReportVerifier {
bool VerifyRTCRemoteOutboundRTPStreamStats(
const RTCRemoteOutboundRtpStreamStats& remote_outbound_stream) {
RTCStatsVerifier verifier(report_, &remote_outbound_stream);
RTCStatsVerifier verifier(report_.get(), &remote_outbound_stream);
VerifyRTCRTPStreamStats(remote_outbound_stream, verifier);
VerifyRTCSentRTPStreamStats(remote_outbound_stream, verifier);
verifier.TestMemberIsIDReference(remote_outbound_stream.local_id,
@ -1088,7 +1088,7 @@ class RTCStatsReportVerifier {
}
bool VerifyRTCAudioSourceStats(const RTCAudioSourceStats& audio_source) {
RTCStatsVerifier verifier(report_, &audio_source);
RTCStatsVerifier verifier(report_.get(), &audio_source);
VerifyRTCMediaSourceStats(audio_source, &verifier);
// Audio level, unlike audio energy, only gets updated at a certain
// frequency, so we don't require that one to be positive to avoid a race
@ -1106,7 +1106,7 @@ class RTCStatsReportVerifier {
}
bool VerifyRTCVideoSourceStats(const RTCVideoSourceStats& video_source) {
RTCStatsVerifier verifier(report_, &video_source);
RTCStatsVerifier verifier(report_.get(), &video_source);
VerifyRTCMediaSourceStats(video_source, &verifier);
// TODO(hbos): This integration test uses fakes that doesn't support
// VideoTrackSourceInterface::Stats. When this is fixed we should
@ -1120,7 +1120,7 @@ class RTCStatsReportVerifier {
}
bool VerifyRTCTransportStats(const RTCTransportStats& transport) {
RTCStatsVerifier verifier(report_, &transport);
RTCStatsVerifier verifier(report_.get(), &transport);
verifier.TestMemberIsNonNegative<uint64_t>(transport.bytes_sent);
verifier.TestMemberIsNonNegative<uint64_t>(transport.packets_sent);
verifier.TestMemberIsNonNegative<uint64_t>(transport.bytes_received);
@ -1240,7 +1240,7 @@ TEST_F(RTCStatsIntegrationTest,
rtc::scoped_refptr<RTCStatsObtainer> stats_obtainer =
RTCStatsObtainer::Create();
caller_->pc()->GetStats(stats_obtainer);
caller_->pc()->GetStats(stats_obtainer.get());
// This will destroy the peer connection.
caller_ = nullptr;
// Any pending stats requests should have completed in the act of destroying
@ -1257,7 +1257,7 @@ TEST_F(RTCStatsIntegrationTest, GetsStatsWhileClosingPeerConnection) {
rtc::scoped_refptr<RTCStatsObtainer> stats_obtainer =
RTCStatsObtainer::Create();
caller_->pc()->GetStats(stats_obtainer);
caller_->pc()->GetStats(stats_obtainer.get());
caller_->pc()->Close();
ASSERT_TRUE(stats_obtainer->report());

3
third_party/libwebrtc/pc/rtp_sender.cc поставляемый
Просмотреть файл

@ -651,7 +651,8 @@ void VideoRtpSender::SetSend() {
break;
}
bool success = worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&] {
return video_media_channel()->SetVideoSend(ssrc_, &options, video_track());
return video_media_channel()->SetVideoSend(ssrc_, &options,
video_track().get());
});
RTC_DCHECK(success);
}

30
third_party/libwebrtc/pc/rtp_sender_receiver_unittest.cc поставляемый
Просмотреть файл

@ -210,7 +210,7 @@ class RtpSenderReceiverTest
FakeVideoTrackSource::Create(is_screencast));
video_track_ =
VideoTrack::Create(kVideoTrackId, source, rtc::Thread::Current());
EXPECT_TRUE(local_stream_->AddTrack(video_track_));
EXPECT_TRUE(local_stream_->AddTrack(video_track_.get()));
}
void CreateAudioRtpSender() { CreateAudioRtpSender(nullptr); }
@ -218,13 +218,13 @@ class RtpSenderReceiverTest
void CreateAudioRtpSender(
const rtc::scoped_refptr<LocalAudioSource>& source) {
audio_track_ = AudioTrack::Create(kAudioTrackId, source);
EXPECT_TRUE(local_stream_->AddTrack(audio_track_));
EXPECT_TRUE(local_stream_->AddTrack(audio_track_.get()));
std::unique_ptr<MockSetStreamsObserver> set_streams_observer =
std::make_unique<MockSetStreamsObserver>();
audio_rtp_sender_ =
AudioRtpSender::Create(worker_thread_, audio_track_->id(), nullptr,
set_streams_observer.get());
ASSERT_TRUE(audio_rtp_sender_->SetTrack(audio_track_));
ASSERT_TRUE(audio_rtp_sender_->SetTrack(audio_track_.get()));
EXPECT_CALL(*set_streams_observer, OnSetStreams());
audio_rtp_sender_->SetStreams({local_stream_->id()});
audio_rtp_sender_->SetMediaChannel(voice_media_channel_);
@ -288,7 +288,7 @@ class RtpSenderReceiverTest
std::make_unique<MockSetStreamsObserver>();
video_rtp_sender_ = VideoRtpSender::Create(
worker_thread_, video_track_->id(), set_streams_observer.get());
ASSERT_TRUE(video_rtp_sender_->SetTrack(video_track_));
ASSERT_TRUE(video_rtp_sender_->SetTrack(video_track_.get()));
EXPECT_CALL(*set_streams_observer, OnSetStreams());
video_rtp_sender_->SetStreams({local_stream_->id()});
video_rtp_sender_->SetMediaChannel(video_media_channel_);
@ -733,7 +733,7 @@ TEST_F(RtpSenderReceiverTest, AudioSenderWithoutTrackAndSsrc) {
AudioTrack::Create(kAudioTrackId, nullptr);
// Track but no SSRC.
EXPECT_TRUE(audio_rtp_sender_->SetTrack(track));
EXPECT_TRUE(audio_rtp_sender_->SetTrack(track.get()));
VerifyVoiceChannelNoInput();
// SSRC but no track.
@ -748,7 +748,7 @@ TEST_F(RtpSenderReceiverTest, VideoSenderWithoutTrackAndSsrc) {
CreateVideoRtpSenderWithNoTrack();
// Track but no SSRC.
EXPECT_TRUE(video_rtp_sender_->SetTrack(video_track_));
EXPECT_TRUE(video_rtp_sender_->SetTrack(video_track_.get()));
VerifyVideoChannelNoInput();
// SSRC but no track.
@ -764,7 +764,7 @@ TEST_F(RtpSenderReceiverTest, AudioSenderEarlyWarmupSsrcThenTrack) {
rtc::scoped_refptr<AudioTrackInterface> track =
AudioTrack::Create(kAudioTrackId, nullptr);
audio_rtp_sender_->SetSsrc(kAudioSsrc);
audio_rtp_sender_->SetTrack(track);
audio_rtp_sender_->SetTrack(track.get());
VerifyVoiceChannelInput();
DestroyAudioRtpSender();
@ -776,7 +776,7 @@ TEST_F(RtpSenderReceiverTest, AudioSenderEarlyWarmupTrackThenSsrc) {
CreateAudioRtpSenderWithNoTrack();
rtc::scoped_refptr<AudioTrackInterface> track =
AudioTrack::Create(kAudioTrackId, nullptr);
audio_rtp_sender_->SetTrack(track);
audio_rtp_sender_->SetTrack(track.get());
audio_rtp_sender_->SetSsrc(kAudioSsrc);
VerifyVoiceChannelInput();
@ -789,7 +789,7 @@ TEST_F(RtpSenderReceiverTest, VideoSenderEarlyWarmupSsrcThenTrack) {
AddVideoTrack();
CreateVideoRtpSenderWithNoTrack();
video_rtp_sender_->SetSsrc(kVideoSsrc);
video_rtp_sender_->SetTrack(video_track_);
video_rtp_sender_->SetTrack(video_track_.get());
VerifyVideoChannelInput();
DestroyVideoRtpSender();
@ -800,7 +800,7 @@ TEST_F(RtpSenderReceiverTest, VideoSenderEarlyWarmupSsrcThenTrack) {
TEST_F(RtpSenderReceiverTest, VideoSenderEarlyWarmupTrackThenSsrc) {
AddVideoTrack();
CreateVideoRtpSenderWithNoTrack();
video_rtp_sender_->SetTrack(video_track_);
video_rtp_sender_->SetTrack(video_track_.get());
video_rtp_sender_->SetSsrc(kVideoSsrc);
VerifyVideoChannelInput();
@ -897,13 +897,13 @@ TEST_F(RtpSenderReceiverTest, AudioSenderCanSetParametersBeforeNegotiation) {
TEST_F(RtpSenderReceiverTest, AudioSenderInitParametersMovedAfterNegotiation) {
audio_track_ = AudioTrack::Create(kAudioTrackId, nullptr);
EXPECT_TRUE(local_stream_->AddTrack(audio_track_));
EXPECT_TRUE(local_stream_->AddTrack(audio_track_.get()));
std::unique_ptr<MockSetStreamsObserver> set_streams_observer =
std::make_unique<MockSetStreamsObserver>();
audio_rtp_sender_ = AudioRtpSender::Create(
worker_thread_, audio_track_->id(), nullptr, set_streams_observer.get());
ASSERT_TRUE(audio_rtp_sender_->SetTrack(audio_track_));
ASSERT_TRUE(audio_rtp_sender_->SetTrack(audio_track_.get()));
EXPECT_CALL(*set_streams_observer, OnSetStreams());
audio_rtp_sender_->SetStreams({local_stream_->id()});
@ -1094,7 +1094,7 @@ TEST_F(RtpSenderReceiverTest, VideoSenderInitParametersMovedAfterNegotiation) {
std::make_unique<MockSetStreamsObserver>();
video_rtp_sender_ = VideoRtpSender::Create(worker_thread_, video_track_->id(),
set_streams_observer.get());
ASSERT_TRUE(video_rtp_sender_->SetTrack(video_track_));
ASSERT_TRUE(video_rtp_sender_->SetTrack(video_track_.get()));
EXPECT_CALL(*set_streams_observer, OnSetStreams());
video_rtp_sender_->SetStreams({local_stream_->id()});
@ -1135,7 +1135,7 @@ TEST_F(RtpSenderReceiverTest,
std::make_unique<MockSetStreamsObserver>();
video_rtp_sender_ = VideoRtpSender::Create(worker_thread_, video_track_->id(),
set_streams_observer.get());
ASSERT_TRUE(video_rtp_sender_->SetTrack(video_track_));
ASSERT_TRUE(video_rtp_sender_->SetTrack(video_track_.get()));
EXPECT_CALL(*set_streams_observer, OnSetStreams());
video_rtp_sender_->SetStreams({local_stream_->id()});
@ -1524,7 +1524,7 @@ TEST_F(RtpSenderReceiverTest,
video_track_->set_content_hint(VideoTrackInterface::ContentHint::kDetailed);
video_rtp_sender_ = VideoRtpSender::Create(worker_thread_, video_track_->id(),
set_streams_observer.get());
ASSERT_TRUE(video_rtp_sender_->SetTrack(video_track_));
ASSERT_TRUE(video_rtp_sender_->SetTrack(video_track_.get()));
EXPECT_CALL(*set_streams_observer, OnSetStreams());
video_rtp_sender_->SetStreams({local_stream_->id()});
video_rtp_sender_->SetMediaChannel(video_media_channel_);

16
third_party/libwebrtc/pc/rtp_transmission_manager.cc поставляемый
Просмотреть файл

@ -177,7 +177,7 @@ RtpTransmissionManager::AddTrackUnifiedPlan(
transceiver->internal()->set_direction(
RtpTransceiverDirection::kSendOnly);
}
transceiver->sender()->SetTrack(track);
transceiver->sender()->SetTrack(track.get());
transceiver->internal()->sender_internal()->set_stream_ids(stream_ids);
transceiver->internal()->set_reused_for_addtrack(true);
} else {
@ -227,7 +227,7 @@ RtpTransmissionManager::CreateSender(
signaling_thread(), VideoRtpSender::Create(worker_thread(), id, this));
NoteUsageEvent(UsageEvent::VIDEO_ADDED);
}
bool set_track_succeeded = sender->SetTrack(track);
bool set_track_succeeded = sender->SetTrack(track.get());
RTC_DCHECK(set_track_succeeded);
sender->internal()->set_stream_ids(stream_ids);
sender->internal()->set_init_send_encodings(send_encodings);
@ -409,7 +409,7 @@ void RtpTransmissionManager::RemoveAudioTrack(AudioTrackInterface* track,
<< " doesn't exist.";
return;
}
GetAudioTransceiver()->internal()->RemoveSender(sender);
GetAudioTransceiver()->internal()->RemoveSender(sender.get());
}
void RtpTransmissionManager::AddVideoTrack(VideoTrackInterface* track,
@ -448,7 +448,7 @@ void RtpTransmissionManager::RemoveVideoTrack(VideoTrackInterface* track,
<< " doesn't exist.";
return;
}
GetVideoTransceiver()->internal()->RemoveSender(sender);
GetVideoTransceiver()->internal()->RemoveSender(sender.get());
}
void RtpTransmissionManager::CreateAudioReceiver(
@ -511,9 +511,9 @@ RtpTransmissionManager::RemoveAndStopReceiver(
return nullptr;
}
if (receiver->media_type() == cricket::MEDIA_TYPE_AUDIO) {
GetAudioTransceiver()->internal()->RemoveReceiver(receiver);
GetAudioTransceiver()->internal()->RemoveReceiver(receiver.get());
} else {
GetVideoTransceiver()->internal()->RemoveReceiver(receiver);
GetVideoTransceiver()->internal()->RemoveReceiver(receiver.get());
}
return receiver;
}
@ -553,7 +553,7 @@ void RtpTransmissionManager::OnRemoteSenderRemoved(
rtc::scoped_refptr<AudioTrackInterface> audio_track =
stream->FindAudioTrack(sender_info.sender_id);
if (audio_track) {
stream->RemoveTrack(audio_track);
stream->RemoveTrack(audio_track.get());
}
} else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
// Stopping or destroying a VideoRtpReceiver will end the
@ -564,7 +564,7 @@ void RtpTransmissionManager::OnRemoteSenderRemoved(
if (video_track) {
// There's no guarantee the track is still available, e.g. the track may
// have been removed from the stream by an application.
stream->RemoveTrack(video_track);
stream->RemoveTrack(video_track.get());
}
} else {
RTC_DCHECK_NOTREACHED() << "Invalid media type";

2
third_party/libwebrtc/pc/scenario_tests/goog_cc_test.cc поставляемый
Просмотреть файл

@ -75,7 +75,7 @@ TEST(GoogCcPeerScenarioTest, MAYBE_NoBweChangeFromVideoUnmute) {
auto get_bwe = [&] {
auto callback =
rtc::make_ref_counted<webrtc::MockRTCStatsCollectorCallback>();
caller->pc()->GetStats(callback);
caller->pc()->GetStats(callback.get());
s.net()->time_controller()->Wait([&] { return callback->called(); });
auto stats =
callback->report()->GetStatsOfType<RTCIceCandidatePairStats>()[0];

40
third_party/libwebrtc/pc/sdp_offer_answer.cc поставляемый
Просмотреть файл

@ -1100,10 +1100,10 @@ class SdpOfferAnswerHandler::SetSessionDescriptionObserverAdapter
return;
if (error.ok()) {
handler_->pc_->message_handler()->PostSetSessionDescriptionSuccess(
inner_observer_);
inner_observer_.get());
} else {
handler_->pc_->message_handler()->PostSetSessionDescriptionFailure(
inner_observer_, std::move(error));
inner_observer_.get(), std::move(error));
}
}
@ -2029,7 +2029,7 @@ void SdpOfferAnswerHandler::ApplyRemoteDescriptionUpdateTransceiverState(
// Once all processing has finished, fire off callbacks.
auto observer = pc_->Observer();
for (const auto& transceiver : now_receiving_transceivers) {
pc_->stats()->AddTrack(transceiver->receiver()->track());
pc_->stats()->AddTrack(transceiver->receiver()->track().get());
observer->OnTrack(transceiver);
observer->OnAddTrack(transceiver->receiver(),
transceiver->receiver()->streams());
@ -2076,7 +2076,7 @@ void SdpOfferAnswerHandler::PlanBUpdateSendersAndReceivers(
RtpTransceiverDirectionHasSend(audio_desc->direction());
UpdateRemoteSendersList(GetActiveStreams(audio_desc),
default_audio_track_needed, audio_desc->type(),
new_streams);
new_streams.get());
}
}
@ -2091,7 +2091,7 @@ void SdpOfferAnswerHandler::PlanBUpdateSendersAndReceivers(
RtpTransceiverDirectionHasSend(video_desc->direction());
UpdateRemoteSendersList(GetActiveStreams(video_desc),
default_video_track_needed, video_desc->type(),
new_streams);
new_streams.get());
}
}
@ -2229,7 +2229,8 @@ void SdpOfferAnswerHandler::DoCreateOffer(
std::string error = "CreateOffer called when PeerConnection is closed.";
RTC_LOG(LS_ERROR) << error;
pc_->message_handler()->PostCreateSessionDescriptionFailure(
observer, RTCError(RTCErrorType::INVALID_STATE, std::move(error)));
observer.get(),
RTCError(RTCErrorType::INVALID_STATE, std::move(error)));
return;
}
@ -2239,7 +2240,7 @@ void SdpOfferAnswerHandler::DoCreateOffer(
std::string error_message = GetSessionErrorMsg();
RTC_LOG(LS_ERROR) << "CreateOffer: " << error_message;
pc_->message_handler()->PostCreateSessionDescriptionFailure(
observer,
observer.get(),
RTCError(RTCErrorType::INTERNAL_ERROR, std::move(error_message)));
return;
}
@ -2248,7 +2249,8 @@ void SdpOfferAnswerHandler::DoCreateOffer(
std::string error = "CreateOffer called with invalid options.";
RTC_LOG(LS_ERROR) << error;
pc_->message_handler()->PostCreateSessionDescriptionFailure(
observer, RTCError(RTCErrorType::INVALID_PARAMETER, std::move(error)));
observer.get(),
RTCError(RTCErrorType::INVALID_PARAMETER, std::move(error)));
return;
}
@ -2258,14 +2260,15 @@ void SdpOfferAnswerHandler::DoCreateOffer(
RTCError error = HandleLegacyOfferOptions(options);
if (!error.ok()) {
pc_->message_handler()->PostCreateSessionDescriptionFailure(
observer, std::move(error));
observer.get(), std::move(error));
return;
}
}
cricket::MediaSessionOptions session_options;
GetOptionsForOffer(options, &session_options);
webrtc_session_desc_factory_->CreateOffer(observer, options, session_options);
webrtc_session_desc_factory_->CreateOffer(observer.get(), options,
session_options);
}
void SdpOfferAnswerHandler::CreateAnswer(
@ -2313,7 +2316,7 @@ void SdpOfferAnswerHandler::DoCreateAnswer(
std::string error_message = GetSessionErrorMsg();
RTC_LOG(LS_ERROR) << "CreateAnswer: " << error_message;
pc_->message_handler()->PostCreateSessionDescriptionFailure(
observer,
observer.get(),
RTCError(RTCErrorType::INTERNAL_ERROR, std::move(error_message)));
return;
}
@ -2325,7 +2328,8 @@ void SdpOfferAnswerHandler::DoCreateAnswer(
"have-remote-offer or have-local-pranswer.";
RTC_LOG(LS_ERROR) << error;
pc_->message_handler()->PostCreateSessionDescriptionFailure(
observer, RTCError(RTCErrorType::INVALID_STATE, std::move(error)));
observer.get(),
RTCError(RTCErrorType::INVALID_STATE, std::move(error)));
return;
}
@ -2349,7 +2353,7 @@ void SdpOfferAnswerHandler::DoCreateAnswer(
cricket::MediaSessionOptions session_options;
GetOptionsForAnswer(options, &session_options);
webrtc_session_desc_factory_->CreateAnswer(observer, session_options);
webrtc_session_desc_factory_->CreateAnswer(observer.get(), session_options);
}
void SdpOfferAnswerHandler::DoSetRemoteDescription(
@ -2775,7 +2779,7 @@ bool SdpOfferAnswerHandler::AddStream(MediaStreamInterface* local_stream) {
if (pc_->IsClosed()) {
return false;
}
if (!CanAddLocalMediaStream(local_streams_, local_stream)) {
if (!CanAddLocalMediaStream(local_streams_.get(), local_stream)) {
return false;
}
@ -4245,7 +4249,7 @@ void SdpOfferAnswerHandler::RemoveRemoteStreamsIfEmpty(
for (const auto& remote_stream : remote_streams) {
if (remote_stream->GetAudioTracks().empty() &&
remote_stream->GetVideoTracks().empty()) {
remote_streams_->RemoveStream(remote_stream);
remote_streams_->RemoveStream(remote_stream.get());
removed_streams->push_back(remote_stream);
}
}
@ -4373,7 +4377,7 @@ void SdpOfferAnswerHandler::UpdateRemoteSendersList(
rtp_manager()->FindSenderInfo(*current_senders, stream_id, sender_id);
if (!sender_info) {
current_senders->push_back(RtpSenderInfo(stream_id, sender_id, ssrc));
rtp_manager()->OnRemoteSenderAdded(current_senders->back(), stream,
rtp_manager()->OnRemoteSenderAdded(current_senders->back(), stream.get(),
media_type);
}
}
@ -4398,7 +4402,7 @@ void SdpOfferAnswerHandler::UpdateRemoteSendersList(
current_senders->push_back(
RtpSenderInfo(kDefaultStreamId, default_sender_id, /*ssrc=*/0));
rtp_manager()->OnRemoteSenderAdded(current_senders->back(),
default_stream, media_type);
default_stream.get(), media_type);
}
}
}
@ -4633,7 +4637,7 @@ void SdpOfferAnswerHandler::UpdateEndedRemoteMediaStreams() {
}
for (auto& stream : streams_to_remove) {
remote_streams_->RemoveStream(stream);
remote_streams_->RemoveStream(stream.get());
pc_->Observer()->OnRemoveStream(std::move(stream));
}
}

2
third_party/libwebrtc/pc/sdp_offer_answer_unittest.cc поставляемый
Просмотреть файл

@ -85,7 +85,7 @@ class SdpOfferAnswerTest : public ::testing::Test {
auto result = pc_factory_->CreatePeerConnectionOrError(
config, PeerConnectionDependencies(observer.get()));
EXPECT_TRUE(result.ok());
observer->SetPeerConnectionInterface(result.value());
observer->SetPeerConnectionInterface(result.value().get());
return std::make_unique<PeerConnectionWrapper>(
pc_factory_, result.MoveValue(), std::move(observer));
}

149
third_party/libwebrtc/pc/stats_collector_unittest.cc поставляемый
Просмотреть файл

@ -666,7 +666,7 @@ class StatsCollectorTest : public ::testing::Test {
const std::string kTransportName = "transport";
auto pc = CreatePeerConnection();
auto stats = CreateStatsCollector(pc);
auto stats = CreateStatsCollector(pc.get());
pc->AddVoiceChannel("audio", kTransportName);
@ -779,10 +779,10 @@ class StatsCollectorTrackTest : public StatsCollectorTest,
if (GetParam()) {
if (!stream_)
stream_ = MediaStream::Create("streamid");
stream_->AddTrack(track_);
stats->AddStream(stream_);
stream_->AddTrack(track_.get());
stats->AddStream(stream_.get());
} else {
stats->AddTrack(track_);
stats->AddTrack(track_.get());
}
pc->AddSender(CreateMockSender(track_, kSsrcOfTrack));
}
@ -794,10 +794,10 @@ class StatsCollectorTrackTest : public StatsCollectorTest,
rtc::Thread::Current());
if (GetParam()) {
stream_ = MediaStream::Create("streamid");
stream_->AddTrack(track_);
stats->AddStream(stream_);
stream_->AddTrack(track_.get());
stats->AddStream(stream_.get());
} else {
stats->AddTrack(track_);
stats->AddTrack(track_.get());
}
pc->AddReceiver(CreateMockReceiver(track_, kSsrcOfTrack));
}
@ -813,10 +813,10 @@ class StatsCollectorTrackTest : public StatsCollectorTest,
if (GetParam()) {
if (!stream_)
stream_ = MediaStream::Create("streamid");
stream_->AddTrack(audio_track_);
stats->AddStream(stream_);
stream_->AddTrack(audio_track_.get());
stats->AddStream(stream_.get());
} else {
stats->AddTrack(audio_track_);
stats->AddTrack(audio_track_.get());
}
return pc->AddSender(CreateMockSender(audio_track_, kSsrcOfTrack));
}
@ -826,12 +826,12 @@ class StatsCollectorTrackTest : public StatsCollectorTest,
StatsCollectorForTest* stats) {
audio_track_ = rtc::make_ref_counted<FakeAudioTrack>(kRemoteTrackId);
if (GetParam()) {
if (stream_ == NULL)
if (stream_ == nullptr)
stream_ = MediaStream::Create("streamid");
stream_->AddTrack(audio_track_);
stats->AddStream(stream_);
stream_->AddTrack(audio_track_.get());
stats->AddStream(stream_.get());
} else {
stats->AddTrack(audio_track_);
stats->AddTrack(audio_track_.get());
}
pc->AddReceiver(CreateMockReceiver(audio_track_, kSsrcOfTrack));
}
@ -843,7 +843,7 @@ class StatsCollectorTrackTest : public StatsCollectorTest,
TEST_F(StatsCollectorTest, FilterOutNegativeDataChannelId) {
auto pc = CreatePeerConnection();
auto stats = CreateStatsCollector(pc);
auto stats = CreateStatsCollector(pc.get());
pc->AddSctpDataChannel("hacks");
@ -867,7 +867,7 @@ TEST_F(StatsCollectorTest, ExtractDataInfo) {
DataChannelInterface::DataState::kConnecting);
auto pc = CreatePeerConnection();
auto stats = CreateStatsCollector(pc);
auto stats = CreateStatsCollector(pc.get());
InternalDataChannelInit init;
init.id = kDataChannelId;
@ -906,7 +906,7 @@ TEST_P(StatsCollectorTrackTest, BytesCounterHandles64Bits) {
constexpr int64_t kBytesSent = 12345678901234LL;
auto pc = CreatePeerConnection();
auto stats = CreateStatsCollector(pc);
auto stats = CreateStatsCollector(pc.get());
VideoSenderInfo video_sender_info;
video_sender_info.add_ssrc(1234);
@ -917,7 +917,7 @@ TEST_P(StatsCollectorTrackTest, BytesCounterHandles64Bits) {
pc->AddVideoChannel("video", "transport", video_info);
AddOutgoingVideoTrack(pc, stats.get());
AddOutgoingVideoTrack(pc.get(), stats.get());
stats->UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
StatsReports reports;
@ -938,7 +938,7 @@ TEST_P(StatsCollectorTrackTest, AudioBandwidthEstimationInfoIsReported) {
constexpr int kPacerDelay = 123;
auto pc = CreatePeerConnection();
auto stats = CreateStatsCollector(pc);
auto stats = CreateStatsCollector(pc.get());
VoiceSenderInfo voice_sender_info;
voice_sender_info.add_ssrc(1234);
@ -950,7 +950,7 @@ TEST_P(StatsCollectorTrackTest, AudioBandwidthEstimationInfoIsReported) {
auto* voice_media_channel = pc->AddVoiceChannel("audio", "transport");
voice_media_channel->SetStats(voice_info);
AddOutgoingAudioTrack(pc, stats.get());
AddOutgoingAudioTrack(pc.get(), stats.get());
Call::Stats call_stats;
call_stats.send_bandwidth_bps = kSendBandwidth;
@ -987,7 +987,7 @@ TEST_P(StatsCollectorTrackTest, VideoBandwidthEstimationInfoIsReported) {
constexpr int kPacerDelay = 123;
auto pc = CreatePeerConnection();
auto stats = CreateStatsCollector(pc);
auto stats = CreateStatsCollector(pc.get());
VideoSenderInfo video_sender_info;
video_sender_info.add_ssrc(1234);
@ -999,7 +999,7 @@ TEST_P(StatsCollectorTrackTest, VideoBandwidthEstimationInfoIsReported) {
pc->AddVideoChannel("video", "transport", video_info);
AddOutgoingVideoTrack(pc, stats.get());
AddOutgoingVideoTrack(pc.get(), stats.get());
Call::Stats call_stats;
call_stats.send_bandwidth_bps = kSendBandwidth;
@ -1030,7 +1030,7 @@ TEST_P(StatsCollectorTrackTest, VideoBandwidthEstimationInfoIsReported) {
// exists in the returned stats.
TEST_F(StatsCollectorTest, SessionObjectExists) {
auto pc = CreatePeerConnection();
auto stats = CreateStatsCollector(pc);
auto stats = CreateStatsCollector(pc.get());
stats->UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
StatsReports reports;
@ -1044,7 +1044,7 @@ TEST_F(StatsCollectorTest, SessionObjectExists) {
// in the returned stats.
TEST_F(StatsCollectorTest, OnlyOneSessionObjectExists) {
auto pc = CreatePeerConnection();
auto stats = CreateStatsCollector(pc);
auto stats = CreateStatsCollector(pc.get());
stats->UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
stats->UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
@ -1061,10 +1061,10 @@ TEST_F(StatsCollectorTest, OnlyOneSessionObjectExists) {
// without calling StatsCollector::UpdateStats.
TEST_P(StatsCollectorTrackTest, TrackObjectExistsWithoutUpdateStats) {
auto pc = CreatePeerConnection();
auto stats = CreateStatsCollector(pc);
auto stats = CreateStatsCollector(pc.get());
pc->AddVideoChannel("video", "transport");
AddOutgoingVideoTrack(pc, stats.get());
AddOutgoingVideoTrack(pc.get(), stats.get());
// Verfies the existence of the track report.
StatsReports reports;
@ -1085,7 +1085,7 @@ TEST_P(StatsCollectorTrackTest, TrackAndSsrcObjectExistAfterUpdateSsrcStats) {
constexpr int64_t kBytesSent = 12345678901234LL;
auto pc = CreatePeerConnection();
auto stats = CreateStatsCollector(pc);
auto stats = CreateStatsCollector(pc.get());
VideoSenderInfo video_sender_info;
video_sender_info.add_ssrc(1234);
@ -1096,7 +1096,7 @@ TEST_P(StatsCollectorTrackTest, TrackAndSsrcObjectExistAfterUpdateSsrcStats) {
pc->AddVideoChannel("video", "transport", video_info);
AddOutgoingVideoTrack(pc, stats.get());
AddOutgoingVideoTrack(pc.get(), stats.get());
stats->UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
StatsReports reports;
@ -1111,7 +1111,7 @@ TEST_P(StatsCollectorTrackTest, TrackAndSsrcObjectExistAfterUpdateSsrcStats) {
// Get report for the specific `track`.
reports.clear();
stats->GetStats(track_, &reports);
stats->GetStats(track_.get(), &reports);
// `reports` should contain at least one session report, one track report,
// and one ssrc report.
EXPECT_LE(3u, reports.size());
@ -1139,7 +1139,7 @@ TEST_P(StatsCollectorTrackTest, TransportObjectLinkedFromSsrcObject) {
constexpr int64_t kBytesSent = 12345678901234LL;
auto pc = CreatePeerConnection();
auto stats = CreateStatsCollector(pc);
auto stats = CreateStatsCollector(pc.get());
VideoSenderInfo video_sender_info;
video_sender_info.add_ssrc(1234);
@ -1150,7 +1150,7 @@ TEST_P(StatsCollectorTrackTest, TransportObjectLinkedFromSsrcObject) {
pc->AddVideoChannel("video", "transport", video_info);
AddOutgoingVideoTrack(pc, stats.get());
AddOutgoingVideoTrack(pc.get(), stats.get());
stats->UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
StatsReports reports;
@ -1181,10 +1181,10 @@ TEST_P(StatsCollectorTrackTest, TransportObjectLinkedFromSsrcObject) {
// an outgoing SSRC where remote stats are not returned.
TEST_P(StatsCollectorTrackTest, RemoteSsrcInfoIsAbsent) {
auto pc = CreatePeerConnection();
auto stats = CreateStatsCollector(pc);
auto stats = CreateStatsCollector(pc.get());
pc->AddVideoChannel("video", "transport");
AddOutgoingVideoTrack(pc, stats.get());
AddOutgoingVideoTrack(pc.get(), stats.get());
stats->UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
StatsReports reports;
@ -1199,7 +1199,7 @@ TEST_P(StatsCollectorTrackTest, RemoteSsrcInfoIsAbsent) {
// an outgoing SSRC where stats are returned.
TEST_P(StatsCollectorTrackTest, RemoteSsrcInfoIsPresent) {
auto pc = CreatePeerConnection();
auto stats = CreateStatsCollector(pc);
auto stats = CreateStatsCollector(pc.get());
SsrcReceiverInfo remote_ssrc_stats;
remote_ssrc_stats.timestamp = 12345.678;
@ -1212,7 +1212,7 @@ TEST_P(StatsCollectorTrackTest, RemoteSsrcInfoIsPresent) {
pc->AddVideoChannel("video", "transport", video_info);
AddOutgoingVideoTrack(pc, stats.get());
AddOutgoingVideoTrack(pc.get(), stats.get());
stats->UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
StatsReports reports;
@ -1230,7 +1230,7 @@ TEST_P(StatsCollectorTrackTest, ReportsFromRemoteTrack) {
constexpr int64_t kNumOfPacketsConcealed = 54321;
auto pc = CreatePeerConnection();
auto stats = CreateStatsCollector(pc);
auto stats = CreateStatsCollector(pc.get());
VideoReceiverInfo video_receiver_info;
video_receiver_info.add_ssrc(1234);
@ -1240,7 +1240,7 @@ TEST_P(StatsCollectorTrackTest, ReportsFromRemoteTrack) {
pc->AddVideoChannel("video", "transport", video_info);
AddIncomingVideoTrack(pc, stats.get());
AddIncomingVideoTrack(pc.get(), stats.get());
stats->UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
StatsReports reports;
@ -1279,7 +1279,7 @@ TEST_F(StatsCollectorTest, IceCandidateReport) {
const rtc::SocketAddress kRemoteAddress(kRemoteIp, kRemotePort);
auto pc = CreatePeerConnection();
auto stats = CreateStatsCollector(pc);
auto stats = CreateStatsCollector(pc.get());
cricket::Candidate local;
EXPECT_GT(local.id().length(), 0u);
@ -1415,7 +1415,7 @@ TEST_F(StatsCollectorTest, ChainlessCertificateReportsCreated) {
// transport is present.
TEST_F(StatsCollectorTest, NoTransport) {
auto pc = CreatePeerConnection();
auto stats = CreateStatsCollector(pc);
auto stats = CreateStatsCollector(pc.get());
// This will cause the fake PeerConnection to generate a TransportStats entry
// but with only a single dummy TransportChannelStats.
@ -1474,16 +1474,16 @@ TEST_P(StatsCollectorTrackTest, FilterOutNegativeInitialValues) {
}
auto pc = CreatePeerConnection();
auto stats = CreateStatsCollector(pc);
auto stats = CreateStatsCollector(pc.get());
// Create a local stream with a local audio track and adds it to the stats.
stream_ = MediaStream::Create("streamid");
auto local_track =
rtc::make_ref_counted<FakeAudioTrackWithInitValue>(kLocalTrackId);
stream_->AddTrack(local_track);
stream_->AddTrack(local_track.get());
pc->AddSender(CreateMockSender(local_track, kSsrcOfTrack));
if (GetParam()) {
stats->AddStream(stream_);
stats->AddStream(stream_.get());
}
stats->AddLocalAudioTrack(local_track.get(), kSsrcOfTrack);
@ -1492,10 +1492,10 @@ TEST_P(StatsCollectorTrackTest, FilterOutNegativeInitialValues) {
MediaStream::Create("remotestreamid"));
auto remote_track =
rtc::make_ref_counted<FakeAudioTrackWithInitValue>(kRemoteTrackId);
remote_stream->AddTrack(remote_track);
remote_stream->AddTrack(remote_track.get());
pc->AddReceiver(CreateMockReceiver(remote_track, kSsrcOfTrack));
if (GetParam()) {
stats->AddStream(remote_stream);
stats->AddStream(remote_stream.get());
}
VoiceSenderInfo voice_sender_info;
@ -1561,14 +1561,15 @@ TEST_P(StatsCollectorTrackTest, FilterOutNegativeInitialValues) {
// AudioTrackInterface::GetStats() method.
TEST_P(StatsCollectorTrackTest, GetStatsFromLocalAudioTrack) {
auto pc = CreatePeerConnection();
auto stats = CreateStatsCollector(pc);
auto stats = CreateStatsCollector(pc.get());
AddOutgoingAudioTrack(pc, stats.get());
stats->AddLocalAudioTrack(audio_track_, kSsrcOfTrack);
AddOutgoingAudioTrack(pc.get(), stats.get());
stats->AddLocalAudioTrack(audio_track_.get(), kSsrcOfTrack);
VoiceSenderInfo voice_sender_info;
InitVoiceSenderInfo(&voice_sender_info);
UpdateVoiceSenderInfoFromAudioTrack(audio_track_, &voice_sender_info, false);
UpdateVoiceSenderInfoFromAudioTrack(audio_track_.get(), &voice_sender_info,
false);
VoiceMediaInfo voice_info;
voice_info.senders.push_back(voice_sender_info);
@ -1576,7 +1577,7 @@ TEST_P(StatsCollectorTrackTest, GetStatsFromLocalAudioTrack) {
voice_media_channel->SetStats(voice_info);
StatsReports reports; // returned values.
VerifyAudioTrackStats(audio_track_, stats.get(), voice_info, &reports);
VerifyAudioTrackStats(audio_track_.get(), stats.get(), voice_info, &reports);
// Verify that there is no remote report for the local audio track because
// we did not set it up.
@ -1589,9 +1590,9 @@ TEST_P(StatsCollectorTrackTest, GetStatsFromLocalAudioTrack) {
// correctly.
TEST_P(StatsCollectorTrackTest, GetStatsFromRemoteStream) {
auto pc = CreatePeerConnection();
auto stats = CreateStatsCollector(pc);
auto stats = CreateStatsCollector(pc.get());
AddIncomingAudioTrack(pc, stats.get());
AddIncomingAudioTrack(pc.get(), stats.get());
VoiceReceiverInfo voice_receiver_info;
InitVoiceReceiverInfo(&voice_receiver_info);
@ -1603,16 +1604,16 @@ TEST_P(StatsCollectorTrackTest, GetStatsFromRemoteStream) {
voice_media_channel->SetStats(voice_info);
StatsReports reports; // returned values.
VerifyAudioTrackStats(audio_track_, stats.get(), voice_info, &reports);
VerifyAudioTrackStats(audio_track_.get(), stats.get(), voice_info, &reports);
}
// This test verifies that a local stats object won't update its statistics
// after a RemoveLocalAudioTrack() call.
TEST_P(StatsCollectorTrackTest, GetStatsAfterRemoveAudioStream) {
auto pc = CreatePeerConnection();
auto stats = CreateStatsCollector(pc);
auto stats = CreateStatsCollector(pc.get());
AddOutgoingAudioTrack(pc, stats.get());
AddOutgoingAudioTrack(pc.get(), stats.get());
stats->AddLocalAudioTrack(audio_track_.get(), kSsrcOfTrack);
VoiceSenderInfo voice_sender_info;
@ -1651,10 +1652,10 @@ TEST_P(StatsCollectorTrackTest, GetStatsAfterRemoveAudioStream) {
// the same ssrc, they populate stats reports correctly.
TEST_P(StatsCollectorTrackTest, LocalAndRemoteTracksWithSameSsrc) {
auto pc = CreatePeerConnection();
auto stats = CreateStatsCollector(pc);
auto stats = CreateStatsCollector(pc.get());
// Create a local stream with a local audio track and adds it to the stats.
AddOutgoingAudioTrack(pc, stats.get());
AddOutgoingAudioTrack(pc.get(), stats.get());
stats->AddLocalAudioTrack(audio_track_.get(), kSsrcOfTrack);
// Create a remote stream with a remote audio track and adds it to the stats.
@ -1662,8 +1663,8 @@ TEST_P(StatsCollectorTrackTest, LocalAndRemoteTracksWithSameSsrc) {
MediaStream::Create("remotestreamid"));
auto remote_track = rtc::make_ref_counted<FakeAudioTrack>(kRemoteTrackId);
pc->AddReceiver(CreateMockReceiver(remote_track, kSsrcOfTrack));
remote_stream->AddTrack(remote_track);
stats->AddStream(remote_stream);
remote_stream->AddTrack(remote_track.get());
stats->AddStream(remote_stream.get());
VoiceSenderInfo voice_sender_info;
InitVoiceSenderInfo(&voice_sender_info);
@ -1723,15 +1724,16 @@ TEST_P(StatsCollectorTrackTest, TwoLocalTracksWithSameSsrc) {
}
auto pc = CreatePeerConnection();
auto stats = CreateStatsCollector(pc);
auto stats = CreateStatsCollector(pc.get());
// Create a local stream with a local audio track and adds it to the stats.
auto sender = AddOutgoingAudioTrack(pc, stats.get());
stats->AddLocalAudioTrack(audio_track_, kSsrcOfTrack);
auto sender = AddOutgoingAudioTrack(pc.get(), stats.get());
stats->AddLocalAudioTrack(audio_track_.get(), kSsrcOfTrack);
VoiceSenderInfo voice_sender_info;
InitVoiceSenderInfo(&voice_sender_info);
UpdateVoiceSenderInfoFromAudioTrack(audio_track_, &voice_sender_info, false);
UpdateVoiceSenderInfoFromAudioTrack(audio_track_.get(), &voice_sender_info,
false);
voice_sender_info.add_ssrc(kSsrcOfTrack);
VoiceMediaInfo voice_info;
voice_info.senders.push_back(voice_sender_info);
@ -1740,7 +1742,7 @@ TEST_P(StatsCollectorTrackTest, TwoLocalTracksWithSameSsrc) {
voice_media_channel->SetStats(voice_info);
StatsReports reports; // returned values.
VerifyAudioTrackStats(audio_track_, stats.get(), voice_info, &reports);
VerifyAudioTrackStats(audio_track_.get(), stats.get(), voice_info, &reports);
// Remove the previous audio track from the stream.
stream_->RemoveTrack(audio_track_.get());
@ -1751,21 +1753,22 @@ TEST_P(StatsCollectorTrackTest, TwoLocalTracksWithSameSsrc) {
static const std::string kNewTrackId = "new_track_id";
auto new_audio_track = rtc::make_ref_counted<FakeAudioTrack>(kNewTrackId);
pc->AddSender(CreateMockSender(new_audio_track, kSsrcOfTrack));
stream_->AddTrack(new_audio_track);
stream_->AddTrack(new_audio_track.get());
stats->AddLocalAudioTrack(new_audio_track, kSsrcOfTrack);
stats->AddLocalAudioTrack(new_audio_track.get(), kSsrcOfTrack);
stats->InvalidateCache();
VoiceSenderInfo new_voice_sender_info;
InitVoiceSenderInfo(&new_voice_sender_info);
UpdateVoiceSenderInfoFromAudioTrack(new_audio_track, &new_voice_sender_info,
false);
UpdateVoiceSenderInfoFromAudioTrack(new_audio_track.get(),
&new_voice_sender_info, false);
VoiceMediaInfo new_voice_info;
new_voice_info.senders.push_back(new_voice_sender_info);
voice_media_channel->SetStats(new_voice_info);
reports.clear();
VerifyAudioTrackStats(new_audio_track, stats.get(), new_voice_info, &reports);
VerifyAudioTrackStats(new_audio_track.get(), stats.get(), new_voice_info,
&reports);
}
// Test that if there are two local senders with the same track then two SSRC
@ -1776,7 +1779,7 @@ TEST_P(StatsCollectorTrackTest, TwoLocalSendersWithSameTrack) {
constexpr uint32_t kSecondSsrc = 33;
auto pc = CreatePeerConnection();
auto stats = CreateStatsCollector(pc);
auto stats = CreateStatsCollector(pc.get());
auto local_track =
rtc::make_ref_counted<FakeAudioTrackWithInitValue>(kLocalTrackId);
@ -1836,9 +1839,9 @@ TEST_P(StatsCollectorTrackTest, TwoLocalSendersWithSameTrack) {
// This test verifies that stats are correctly set in video send ssrc stats.
TEST_P(StatsCollectorTrackTest, VerifyVideoSendSsrcStats) {
auto pc = CreatePeerConnection();
auto stats = CreateStatsCollector(pc);
auto stats = CreateStatsCollector(pc.get());
AddOutgoingVideoTrack(pc, stats.get());
AddOutgoingVideoTrack(pc.get(), stats.get());
VideoSenderInfo video_sender_info;
video_sender_info.add_ssrc(1234);
@ -1863,9 +1866,9 @@ TEST_P(StatsCollectorTrackTest, VerifyVideoSendSsrcStats) {
// This test verifies that stats are correctly set in video receive ssrc stats.
TEST_P(StatsCollectorTrackTest, VerifyVideoReceiveSsrcStatsNew) {
auto pc = CreatePeerConnection();
auto stats = CreateStatsCollector(pc);
auto stats = CreateStatsCollector(pc.get());
AddIncomingVideoTrack(pc, stats.get());
AddIncomingVideoTrack(pc.get(), stats.get());
VideoReceiverInfo video_receiver_info;
video_receiver_info.add_ssrc(1234);

10
third_party/libwebrtc/pc/stream_collection.h поставляемый
Просмотреть файл

@ -34,14 +34,14 @@ class StreamCollection : public StreamCollectionInterface {
virtual size_t count() { return media_streams_.size(); }
virtual MediaStreamInterface* at(size_t index) {
return media_streams_.at(index);
return media_streams_.at(index).get();
}
virtual MediaStreamInterface* find(const std::string& id) {
for (StreamVector::iterator it = media_streams_.begin();
it != media_streams_.end(); ++it) {
if ((*it)->id().compare(id) == 0) {
return (*it);
return (*it).get();
}
}
return NULL;
@ -49,7 +49,8 @@ class StreamCollection : public StreamCollectionInterface {
virtual MediaStreamTrackInterface* FindAudioTrack(const std::string& id) {
for (size_t i = 0; i < media_streams_.size(); ++i) {
MediaStreamTrackInterface* track = media_streams_[i]->FindAudioTrack(id);
MediaStreamTrackInterface* track =
media_streams_[i]->FindAudioTrack(id).get();
if (track) {
return track;
}
@ -59,7 +60,8 @@ class StreamCollection : public StreamCollectionInterface {
virtual MediaStreamTrackInterface* FindVideoTrack(const std::string& id) {
for (size_t i = 0; i < media_streams_.size(); ++i) {
MediaStreamTrackInterface* track = media_streams_[i]->FindVideoTrack(id);
MediaStreamTrackInterface* track =
media_streams_[i]->FindVideoTrack(id).get();
if (track) {
return track;
}

4
third_party/libwebrtc/pc/test/fake_peer_connection_for_stats.h поставляемый
Просмотреть файл

@ -180,7 +180,7 @@ class FakePeerConnectionForStats : public FakePeerConnectionBase {
void RemoveSender(rtc::scoped_refptr<RtpSenderInterface> sender) {
GetOrCreateFirstTransceiverOfType(sender->media_type())
->internal()
->RemoveSender(sender);
->RemoveSender(sender.get());
}
rtc::scoped_refptr<RtpReceiverInterface> AddReceiver(
@ -198,7 +198,7 @@ class FakePeerConnectionForStats : public FakePeerConnectionBase {
void RemoveReceiver(rtc::scoped_refptr<RtpReceiverInterface> receiver) {
GetOrCreateFirstTransceiverOfType(receiver->media_type())
->internal()
->RemoveReceiver(receiver);
->RemoveReceiver(receiver.get());
}
FakeVoiceMediaChannelForStats* AddVoiceChannel(

36
third_party/libwebrtc/pc/test/integration_test_helpers.h поставляемый
Просмотреть файл

@ -346,7 +346,7 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
// TODO(perkj): Test audio source when it is implemented. Currently audio
// always use the default input.
return peer_connection_factory_->CreateAudioTrack(rtc::CreateRandomUuid(),
source);
source.get());
}
rtc::scoped_refptr<webrtc::VideoTrackInterface> CreateLocalVideoTrack() {
@ -415,7 +415,7 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
data_channels_.push_back(data_channel_or_error.MoveValue());
ASSERT_TRUE(data_channels_.back().get() != nullptr);
data_observers_.push_back(
std::make_unique<MockDataChannelObserver>(data_channels_.back()));
std::make_unique<MockDataChannelObserver>(data_channels_.back().get()));
}
// Return the last observed data channel.
@ -423,7 +423,7 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
if (data_channels_.size() == 0) {
return nullptr;
}
return data_channels_.back();
return data_channels_.back().get();
}
// Return all data channels.
const std::vector<rtc::scoped_refptr<DataChannelInterface>>& data_channels() {
@ -466,7 +466,8 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
webrtc::MediaStreamTrackInterface* track) {
auto observer = rtc::make_ref_counted<MockStatsObserver>();
EXPECT_TRUE(peer_connection_->GetStats(
observer, nullptr, PeerConnectionInterface::kStatsOutputLevelStandard));
observer.get(), nullptr,
PeerConnectionInterface::kStatsOutputLevelStandard));
EXPECT_TRUE_WAIT(observer->called(), kDefaultTimeout);
return observer;
}
@ -481,7 +482,7 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
rtc::scoped_refptr<const webrtc::RTCStatsReport> NewGetStats() {
auto callback =
rtc::make_ref_counted<webrtc::MockRTCStatsCollectorCallback>();
peer_connection_->GetStats(callback);
peer_connection_->GetStats(callback.get());
EXPECT_TRUE_WAIT(callback->called(), kDefaultTimeout);
return callback->report();
}
@ -542,7 +543,7 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
ADD_FAILURE();
return nullptr;
}
return pc()->remote_streams();
return pc()->remote_streams().get();
}
StreamCollectionInterface* local_streams() {
@ -550,7 +551,7 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
ADD_FAILURE();
return nullptr;
}
return pc()->local_streams();
return pc()->local_streams().get();
}
webrtc::PeerConnectionInterface::SignalingState signaling_state() {
@ -619,8 +620,8 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
std::unique_ptr<SessionDescriptionInterface> CreateOfferAndWait() {
auto observer =
rtc::make_ref_counted<MockCreateSessionDescriptionObserver>();
pc()->CreateOffer(observer, offer_answer_options_);
return WaitForDescriptionFromObserver(observer);
pc()->CreateOffer(observer.get(), offer_answer_options_);
return WaitForDescriptionFromObserver(observer.get());
}
bool Rollback() {
return SetRemoteDescription(
@ -856,9 +857,10 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
config, false /* remote */));
rtc::scoped_refptr<webrtc::VideoTrackInterface> track(
peer_connection_factory_->CreateVideoTrack(
rtc::CreateRandomUuid(), video_track_sources_.back()));
rtc::CreateRandomUuid(), video_track_sources_.back().get()));
if (!local_video_renderer_) {
local_video_renderer_.reset(new webrtc::FakeVideoTrackRenderer(track));
local_video_renderer_.reset(
new webrtc::FakeVideoTrackRenderer(track.get()));
}
return track;
}
@ -900,8 +902,8 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
std::unique_ptr<SessionDescriptionInterface> CreateAnswer() {
auto observer =
rtc::make_ref_counted<MockCreateSessionDescriptionObserver>();
pc()->CreateAnswer(observer, offer_answer_options_);
return WaitForDescriptionFromObserver(observer);
pc()->CreateAnswer(observer.get(), offer_answer_options_);
return WaitForDescriptionFromObserver(observer.get());
}
std::unique_ptr<SessionDescriptionInterface> WaitForDescriptionFromObserver(
@ -930,7 +932,7 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
std::string sdp;
EXPECT_TRUE(desc->ToString(&sdp));
RTC_LOG(LS_INFO) << debug_name_ << ": local SDP contents=\n" << sdp;
pc()->SetLocalDescription(observer, desc.release());
pc()->SetLocalDescription(observer.get(), desc.release());
RemoveUnusedVideoRenderers();
// As mentioned above, we need to send the message immediately after
// SetLocalDescription.
@ -942,7 +944,7 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
bool SetRemoteDescription(std::unique_ptr<SessionDescriptionInterface> desc) {
auto observer = rtc::make_ref_counted<MockSetSessionDescriptionObserver>();
RTC_LOG(LS_INFO) << debug_name_ << ": SetRemoteDescription";
pc()->SetRemoteDescription(observer, desc.release());
pc()->SetRemoteDescription(observer.get(), desc.release());
RemoveUnusedVideoRenderers();
EXPECT_TRUE_WAIT(observer->called(), kDefaultTimeout);
return observer->result();
@ -1056,7 +1058,7 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
ASSERT_TRUE(fake_video_renderers_.find(video_track->id()) ==
fake_video_renderers_.end());
fake_video_renderers_[video_track->id()] =
std::make_unique<FakeVideoTrackRenderer>(video_track);
std::make_unique<FakeVideoTrackRenderer>(video_track.get());
}
}
void OnRemoveTrack(
@ -1139,7 +1141,7 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
RTC_LOG(LS_INFO) << debug_name_ << ": OnDataChannel";
data_channels_.push_back(data_channel);
data_observers_.push_back(
std::make_unique<MockDataChannelObserver>(data_channel));
std::make_unique<MockDataChannelObserver>(data_channel.get()));
}
std::string debug_name_;

6
third_party/libwebrtc/pc/test/mock_peer_connection_observers.h поставляемый
Просмотреть файл

@ -74,7 +74,9 @@ class MockPeerConnectionObserver : public PeerConnectionObserver {
MediaStreamInterface* RemoteStream(const std::string& label) {
return remote_streams_->find(label);
}
StreamCollectionInterface* remote_streams() const { return remote_streams_; }
StreamCollectionInterface* remote_streams() const {
return remote_streams_.get();
}
void OnAddStream(rtc::scoped_refptr<MediaStreamInterface> stream) override {
last_added_stream_ = stream;
remote_streams_->AddStream(stream);
@ -82,7 +84,7 @@ class MockPeerConnectionObserver : public PeerConnectionObserver {
void OnRemoveStream(
rtc::scoped_refptr<MediaStreamInterface> stream) override {
last_removed_stream_ = stream;
remote_streams_->RemoveStream(stream);
remote_streams_->RemoveStream(stream.get());
}
void OnRenegotiationNeeded() override { renegotiation_needed_ = true; }
void OnNegotiationNeededEvent(uint32_t event_id) override {

17
third_party/libwebrtc/pc/test/peer_connection_test_wrapper.cc поставляемый
Просмотреть файл

@ -106,7 +106,7 @@ bool PeerConnectionTestWrapper::CreatePc(
RTC_DCHECK_RUN_ON(&pc_thread_checker_);
fake_audio_capture_module_ = FakeAudioCaptureModule::Create();
if (fake_audio_capture_module_ == NULL) {
if (fake_audio_capture_module_ == nullptr) {
return false;
}
@ -184,7 +184,7 @@ void PeerConnectionTestWrapper::OnIceCandidate(
void PeerConnectionTestWrapper::OnDataChannel(
rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel) {
SignalOnDataChannel(data_channel);
SignalOnDataChannel(data_channel.get());
}
void PeerConnectionTestWrapper::OnSuccess(SessionDescriptionInterface* desc) {
@ -237,7 +237,7 @@ void PeerConnectionTestWrapper::SetLocalDescription(SdpType type,
auto observer = rtc::make_ref_counted<MockSetSessionDescriptionObserver>();
peer_connection_->SetLocalDescription(
observer, webrtc::CreateSessionDescription(type, sdp).release());
observer.get(), webrtc::CreateSessionDescription(type, sdp).release());
}
void PeerConnectionTestWrapper::SetRemoteDescription(SdpType type,
@ -248,7 +248,7 @@ void PeerConnectionTestWrapper::SetRemoteDescription(SdpType type,
auto observer = rtc::make_ref_counted<MockSetSessionDescriptionObserver>();
peer_connection_->SetRemoteDescription(
observer, webrtc::CreateSessionDescription(type, sdp).release());
observer.get(), webrtc::CreateSessionDescription(type, sdp).release());
}
void PeerConnectionTestWrapper::AddIceCandidate(const std::string& sdp_mid,
@ -333,8 +333,8 @@ PeerConnectionTestWrapper::GetUserMedia(
peer_connection_factory_->CreateAudioSource(options);
rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
peer_connection_factory_->CreateAudioTrack(kAudioTrackLabelBase,
source));
stream->AddTrack(audio_track);
source.get()));
stream->AddTrack(audio_track.get());
}
if (video) {
@ -348,9 +348,10 @@ PeerConnectionTestWrapper::GetUserMedia(
std::string videotrack_label = stream_id + kVideoTrackLabelBase;
rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track(
peer_connection_factory_->CreateVideoTrack(videotrack_label, source));
peer_connection_factory_->CreateVideoTrack(videotrack_label,
source.get()));
stream->AddTrack(video_track);
stream->AddTrack(video_track.get());
}
return stream;
}

8
third_party/libwebrtc/pc/track_media_info_map.cc поставляемый
Просмотреть файл

@ -52,7 +52,7 @@ void GetAudioAndVideoTrackBySsrc(
RTC_DCHECK(remote_video_track_by_ssrc->empty());
for (const auto& rtp_sender : rtp_senders) {
cricket::MediaType media_type = rtp_sender->media_type();
MediaStreamTrackInterface* track = rtp_sender->track();
MediaStreamTrackInterface* track = rtp_sender->track().get();
if (!track) {
continue;
}
@ -74,7 +74,7 @@ void GetAudioAndVideoTrackBySsrc(
}
for (const auto& rtp_receiver : rtp_receivers) {
cricket::MediaType media_type = rtp_receiver->media_type();
MediaStreamTrackInterface* track = rtp_receiver->track();
MediaStreamTrackInterface* track = rtp_receiver->track().get();
RTC_DCHECK(track);
RtpParameters params = rtp_receiver->GetParameters();
for (const RtpEncodingParameters& encoding : params.encodings) {
@ -126,10 +126,10 @@ TrackMediaInfoMap::TrackMediaInfoMap(
&unsignaled_video_track);
for (const auto& sender : rtp_senders) {
attachment_id_by_track_[sender->track()] = sender->AttachmentId();
attachment_id_by_track_[sender->track().get()] = sender->AttachmentId();
}
for (const auto& receiver : rtp_receivers) {
attachment_id_by_track_[receiver->track()] = receiver->AttachmentId();
attachment_id_by_track_[receiver->track().get()] = receiver->AttachmentId();
}
if (voice_media_info_) {

83
third_party/libwebrtc/pc/track_media_info_map_unittest.cc поставляемый
Просмотреть файл

@ -205,10 +205,10 @@ class TrackMediaInfoMapTest : public ::testing::Test {
} // namespace
TEST_F(TrackMediaInfoMapTest, SingleSenderReceiverPerTrackWithOneSsrc) {
AddRtpSenderWithSsrcs({1}, local_audio_track_);
AddRtpReceiverWithSsrcs({2}, remote_audio_track_);
AddRtpSenderWithSsrcs({3}, local_video_track_);
AddRtpReceiverWithSsrcs({4}, remote_video_track_);
AddRtpSenderWithSsrcs({1}, local_audio_track_.get());
AddRtpReceiverWithSsrcs({2}, remote_audio_track_.get());
AddRtpSenderWithSsrcs({3}, local_video_track_.get());
AddRtpReceiverWithSsrcs({4}, remote_video_track_.get());
CreateMap();
// Local audio track <-> RTP audio sender
@ -241,10 +241,10 @@ TEST_F(TrackMediaInfoMapTest, SingleSenderReceiverPerTrackWithOneSsrc) {
}
TEST_F(TrackMediaInfoMapTest, SingleSenderReceiverPerTrackWithMissingSsrc) {
AddRtpSenderWithSsrcs({}, local_audio_track_);
AddRtpSenderWithSsrcs({}, local_video_track_);
AddRtpReceiverWithSsrcs({}, remote_audio_track_);
AddRtpReceiverWithSsrcs({}, remote_video_track_);
AddRtpSenderWithSsrcs({}, local_audio_track_.get());
AddRtpSenderWithSsrcs({}, local_video_track_.get());
AddRtpReceiverWithSsrcs({}, remote_audio_track_.get());
AddRtpReceiverWithSsrcs({}, remote_video_track_.get());
CreateMap();
EXPECT_FALSE(map_->GetVoiceSenderInfos(*local_audio_track_));
@ -255,10 +255,10 @@ TEST_F(TrackMediaInfoMapTest, SingleSenderReceiverPerTrackWithMissingSsrc) {
TEST_F(TrackMediaInfoMapTest,
SingleSenderReceiverPerTrackWithAudioAndVideoUseSameSsrc) {
AddRtpSenderWithSsrcs({1}, local_audio_track_);
AddRtpReceiverWithSsrcs({2}, remote_audio_track_);
AddRtpSenderWithSsrcs({1}, local_video_track_);
AddRtpReceiverWithSsrcs({2}, remote_video_track_);
AddRtpSenderWithSsrcs({1}, local_audio_track_.get());
AddRtpReceiverWithSsrcs({2}, remote_audio_track_.get());
AddRtpSenderWithSsrcs({1}, local_video_track_.get());
AddRtpReceiverWithSsrcs({2}, remote_video_track_.get());
CreateMap();
// Local audio track <-> RTP audio sender
@ -291,8 +291,8 @@ TEST_F(TrackMediaInfoMapTest,
}
TEST_F(TrackMediaInfoMapTest, SingleMultiSsrcSenderPerTrack) {
AddRtpSenderWithSsrcs({1, 2}, local_audio_track_);
AddRtpSenderWithSsrcs({3, 4}, local_video_track_);
AddRtpSenderWithSsrcs({1, 2}, local_audio_track_.get());
AddRtpSenderWithSsrcs({3, 4}, local_video_track_.get());
CreateMap();
// Local audio track <-> RTP audio senders
@ -313,10 +313,10 @@ TEST_F(TrackMediaInfoMapTest, SingleMultiSsrcSenderPerTrack) {
}
TEST_F(TrackMediaInfoMapTest, MultipleOneSsrcSendersPerTrack) {
AddRtpSenderWithSsrcs({1}, local_audio_track_);
AddRtpSenderWithSsrcs({2}, local_audio_track_);
AddRtpSenderWithSsrcs({3}, local_video_track_);
AddRtpSenderWithSsrcs({4}, local_video_track_);
AddRtpSenderWithSsrcs({1}, local_audio_track_.get());
AddRtpSenderWithSsrcs({2}, local_audio_track_.get());
AddRtpSenderWithSsrcs({3}, local_video_track_.get());
AddRtpSenderWithSsrcs({4}, local_video_track_.get());
CreateMap();
// Local audio track <-> RTP audio senders
@ -343,10 +343,10 @@ TEST_F(TrackMediaInfoMapTest, MultipleOneSsrcSendersPerTrack) {
}
TEST_F(TrackMediaInfoMapTest, MultipleMultiSsrcSendersPerTrack) {
AddRtpSenderWithSsrcs({1, 2}, local_audio_track_);
AddRtpSenderWithSsrcs({3, 4}, local_audio_track_);
AddRtpSenderWithSsrcs({5, 6}, local_video_track_);
AddRtpSenderWithSsrcs({7, 8}, local_video_track_);
AddRtpSenderWithSsrcs({1, 2}, local_audio_track_.get());
AddRtpSenderWithSsrcs({3, 4}, local_audio_track_.get());
AddRtpSenderWithSsrcs({5, 6}, local_video_track_.get());
AddRtpSenderWithSsrcs({7, 8}, local_video_track_.get());
CreateMap();
// Local audio track <-> RTP audio senders
@ -374,10 +374,10 @@ TEST_F(TrackMediaInfoMapTest, MultipleMultiSsrcSendersPerTrack) {
// SSRCs can be reused for send and receive in loopback.
TEST_F(TrackMediaInfoMapTest, SingleSenderReceiverPerTrackWithSsrcNotUnique) {
AddRtpSenderWithSsrcs({1}, local_audio_track_);
AddRtpReceiverWithSsrcs({1}, remote_audio_track_);
AddRtpSenderWithSsrcs({2}, local_video_track_);
AddRtpReceiverWithSsrcs({2}, remote_video_track_);
AddRtpSenderWithSsrcs({1}, local_audio_track_.get());
AddRtpReceiverWithSsrcs({1}, remote_audio_track_.get());
AddRtpSenderWithSsrcs({2}, local_video_track_.get());
AddRtpReceiverWithSsrcs({2}, remote_video_track_.get());
CreateMap();
// Local audio track <-> RTP audio senders
@ -410,10 +410,10 @@ TEST_F(TrackMediaInfoMapTest, SingleSenderReceiverPerTrackWithSsrcNotUnique) {
}
TEST_F(TrackMediaInfoMapTest, SsrcLookupFunction) {
AddRtpSenderWithSsrcs({1}, local_audio_track_);
AddRtpReceiverWithSsrcs({2}, remote_audio_track_);
AddRtpSenderWithSsrcs({3}, local_video_track_);
AddRtpReceiverWithSsrcs({4}, remote_video_track_);
AddRtpSenderWithSsrcs({1}, local_audio_track_.get());
AddRtpReceiverWithSsrcs({2}, remote_audio_track_.get());
AddRtpSenderWithSsrcs({3}, local_video_track_.get());
AddRtpReceiverWithSsrcs({4}, remote_video_track_.get());
CreateMap();
EXPECT_TRUE(map_->GetVoiceSenderInfoBySsrc(1));
EXPECT_TRUE(map_->GetVoiceReceiverInfoBySsrc(2));
@ -424,11 +424,12 @@ TEST_F(TrackMediaInfoMapTest, SsrcLookupFunction) {
}
TEST_F(TrackMediaInfoMapTest, GetAttachmentIdByTrack) {
AddRtpSenderWithSsrcs({1}, local_audio_track_);
AddRtpSenderWithSsrcs({1}, local_audio_track_.get());
CreateMap();
EXPECT_EQ(rtp_senders_[0]->AttachmentId(),
map_->GetAttachmentIdByTrack(local_audio_track_));
EXPECT_EQ(absl::nullopt, map_->GetAttachmentIdByTrack(local_video_track_));
map_->GetAttachmentIdByTrack(local_audio_track_.get()));
EXPECT_EQ(absl::nullopt,
map_->GetAttachmentIdByTrack(local_video_track_.get()));
}
// Death tests.
@ -442,18 +443,18 @@ class TrackMediaInfoMapDeathTest : public TrackMediaInfoMapTest {
};
TEST_F(TrackMediaInfoMapDeathTest, MultipleOneSsrcReceiversPerTrack) {
AddRtpReceiverWithSsrcs({1}, remote_audio_track_);
AddRtpReceiverWithSsrcs({2}, remote_audio_track_);
AddRtpReceiverWithSsrcs({3}, remote_video_track_);
AddRtpReceiverWithSsrcs({4}, remote_video_track_);
AddRtpReceiverWithSsrcs({1}, remote_audio_track_.get());
AddRtpReceiverWithSsrcs({2}, remote_audio_track_.get());
AddRtpReceiverWithSsrcs({3}, remote_video_track_.get());
AddRtpReceiverWithSsrcs({4}, remote_video_track_.get());
EXPECT_DEATH(CreateMap(), "");
}
TEST_F(TrackMediaInfoMapDeathTest, MultipleMultiSsrcReceiversPerTrack) {
AddRtpReceiverWithSsrcs({1, 2}, remote_audio_track_);
AddRtpReceiverWithSsrcs({3, 4}, remote_audio_track_);
AddRtpReceiverWithSsrcs({5, 6}, remote_video_track_);
AddRtpReceiverWithSsrcs({7, 8}, remote_video_track_);
AddRtpReceiverWithSsrcs({1, 2}, remote_audio_track_.get());
AddRtpReceiverWithSsrcs({3, 4}, remote_audio_track_.get());
AddRtpReceiverWithSsrcs({5, 6}, remote_video_track_.get());
AddRtpReceiverWithSsrcs({7, 8}, remote_video_track_.get());
EXPECT_DEATH(CreateMap(), "");
}

4
third_party/libwebrtc/pc/video_rtp_receiver.cc поставляемый
Просмотреть файл

@ -221,7 +221,7 @@ void VideoRtpReceiver::SetStreams(
}
}
if (removed) {
existing_stream->RemoveTrack(track_);
existing_stream->RemoveTrack(track_.get());
}
}
// Add remote track to any streams that are new.
@ -235,7 +235,7 @@ void VideoRtpReceiver::SetStreams(
}
}
if (added) {
stream->AddTrack(track_);
stream->AddTrack(track_.get());
}
}
streams_ = streams;

14
third_party/libwebrtc/pc/webrtc_session_description_factory.cc поставляемый
Просмотреть файл

@ -171,7 +171,7 @@ WebRtcSessionDescriptionFactory::WebRtcSessionDescriptionFactory(
// `SignalCertificateReady`.
signaling_thread_->Post(
RTC_FROM_HERE, this, MSG_USE_CONSTRUCTOR_CERTIFICATE,
new rtc::ScopedRefMessageData<rtc::RTCCertificate>(certificate));
new rtc::ScopedRefMessageData<rtc::RTCCertificate>(certificate.get()));
} else {
// Generate certificate.
certificate_request_state_ = CERTIFICATE_WAITING;
@ -350,7 +350,7 @@ void WebRtcSessionDescriptionFactory::InternalCreateOffer(
? sdp_info_->local_description()->description()
: nullptr);
if (!desc) {
PostCreateSessionDescriptionFailed(request.observer,
PostCreateSessionDescriptionFailed(request.observer.get(),
"Failed to initialize the offer.");
return;
}
@ -377,7 +377,8 @@ void WebRtcSessionDescriptionFactory::InternalCreateOffer(
}
}
}
PostCreateSessionDescriptionSucceeded(request.observer, std::move(offer));
PostCreateSessionDescriptionSucceeded(request.observer.get(),
std::move(offer));
}
void WebRtcSessionDescriptionFactory::InternalCreateAnswer(
@ -411,7 +412,7 @@ void WebRtcSessionDescriptionFactory::InternalCreateAnswer(
? sdp_info_->local_description()->description()
: nullptr);
if (!desc) {
PostCreateSessionDescriptionFailed(request.observer,
PostCreateSessionDescriptionFailed(request.observer.get(),
"Failed to initialize the answer.");
return;
}
@ -438,7 +439,8 @@ void WebRtcSessionDescriptionFactory::InternalCreateAnswer(
}
}
}
PostCreateSessionDescriptionSucceeded(request.observer, std::move(answer));
PostCreateSessionDescriptionSucceeded(request.observer.get(),
std::move(answer));
}
void WebRtcSessionDescriptionFactory::FailPendingRequests(
@ -448,7 +450,7 @@ void WebRtcSessionDescriptionFactory::FailPendingRequests(
const CreateSessionDescriptionRequest& request =
create_session_description_requests_.front();
PostCreateSessionDescriptionFailed(
request.observer,
request.observer.get(),
((request.type == CreateSessionDescriptionRequest::kOffer)
? "CreateOffer"
: "CreateAnswer") +