Remove not-needed webrtc:: prefixes in pc/
This test drives the new tools_webrtc/remove_extra_namespace.py tool.
Bug: None
Change-Id: I9b590aa1213e4cace2d64d555f4dafd893f03606
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327021
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41141}
diff --git a/pc/audio_rtp_receiver.cc b/pc/audio_rtp_receiver.cc
index a8659de..6e7ca6d 100644
--- a/pc/audio_rtp_receiver.cc
+++ b/pc/audio_rtp_receiver.cc
@@ -278,7 +278,7 @@
}
void AudioRtpReceiver::SetDepacketizerToDecoderFrameTransformer(
- rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) {
+ rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) {
RTC_DCHECK_RUN_ON(worker_thread_);
if (media_channel_) {
media_channel_->SetDepacketizerToDecoderFrameTransformer(
diff --git a/pc/audio_rtp_receiver.h b/pc/audio_rtp_receiver.h
index 86c42d5..36cbdff 100644
--- a/pc/audio_rtp_receiver.h
+++ b/pc/audio_rtp_receiver.h
@@ -118,8 +118,7 @@
std::vector<RtpSource> GetSources() const override;
int AttachmentId() const override { return attachment_id_; }
void SetDepacketizerToDecoderFrameTransformer(
- rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
- override;
+ rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) override;
private:
void RestartMediaChannel(absl::optional<uint32_t> ssrc)
diff --git a/pc/audio_rtp_receiver_unittest.cc b/pc/audio_rtp_receiver_unittest.cc
index 9eb20c9..e031f90 100644
--- a/pc/audio_rtp_receiver_unittest.cc
+++ b/pc/audio_rtp_receiver_unittest.cc
@@ -98,7 +98,7 @@
// thread when a media channel pointer is passed to the receiver via the
// constructor.
TEST(AudioRtpReceiver, OnChangedNotificationsAfterConstruction) {
- webrtc::test::RunLoop loop;
+ test::RunLoop loop;
auto* thread = rtc::Thread::Current(); // Points to loop's thread.
cricket::MockVoiceMediaReceiveChannelInterface receive_channel;
auto receiver = rtc::make_ref_counted<AudioRtpReceiver>(
diff --git a/pc/audio_track.h b/pc/audio_track.h
index ae326b3..92c3141 100644
--- a/pc/audio_track.h
+++ b/pc/audio_track.h
@@ -58,7 +58,7 @@
private:
const rtc::scoped_refptr<AudioSourceInterface> audio_source_;
- RTC_NO_UNIQUE_ADDRESS webrtc::SequenceChecker signaling_thread_checker_;
+ RTC_NO_UNIQUE_ADDRESS SequenceChecker signaling_thread_checker_;
};
} // namespace webrtc
diff --git a/pc/connection_context.h b/pc/connection_context.h
index 399e7c2..af5b7a9 100644
--- a/pc/connection_context.h
+++ b/pc/connection_context.h
@@ -138,7 +138,7 @@
RTC_GUARDED_BY(signaling_thread_);
std::unique_ptr<rtc::NetworkManager> default_network_manager_
RTC_GUARDED_BY(signaling_thread_);
- std::unique_ptr<webrtc::CallFactoryInterface> const call_factory_
+ std::unique_ptr<CallFactoryInterface> const call_factory_
RTC_GUARDED_BY(worker_thread());
std::unique_ptr<rtc::PacketSocketFactory> default_socket_factory_
diff --git a/pc/data_channel_controller_unittest.cc b/pc/data_channel_controller_unittest.cc
index 3b8adb6..7d4e604 100644
--- a/pc/data_channel_controller_unittest.cc
+++ b/pc/data_channel_controller_unittest.cc
@@ -27,7 +27,7 @@
using ::testing::NiceMock;
using ::testing::Return;
-class MockDataChannelTransport : public webrtc::DataChannelTransportInterface {
+class MockDataChannelTransport : public DataChannelTransportInterface {
public:
~MockDataChannelTransport() override {}
diff --git a/pc/data_channel_integrationtest.cc b/pc/data_channel_integrationtest.cc
index faec76d..5a8004c 100644
--- a/pc/data_channel_integrationtest.cc
+++ b/pc/data_channel_integrationtest.cc
@@ -90,7 +90,7 @@
// Some things use a time of "0" as a special value, so we need to start out
// the fake clock at a nonzero time.
// TODO(deadbeef): Fix this.
- AdvanceTime(webrtc::TimeDelta::Seconds(1));
+ AdvanceTime(TimeDelta::Seconds(1));
}
// Explicit handle.
@@ -422,7 +422,7 @@
TEST_P(DataChannelIntegrationTest, SctpDataChannelConfigSentToOtherSide) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
- webrtc::DataChannelInit init;
+ DataChannelInit init;
init.id = 53;
init.maxRetransmits = 52;
caller()->CreateDataChannel("data-channel", &init);
@@ -453,7 +453,7 @@
// Normal procedure, but with unordered data channel config.
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
- webrtc::DataChannelInit init;
+ DataChannelInit init;
init.ordered = false;
caller()->CreateDataChannel(&init);
caller()->CreateAndSetAndSignalOffer();
@@ -515,7 +515,7 @@
const size_t kIterations = 10;
bool has_negotiated = false;
- webrtc::DataChannelInit init;
+ DataChannelInit init;
for (size_t repeats = 0; repeats < kIterations; ++repeats) {
RTC_LOG(LS_INFO) << "Iteration " << (repeats + 1) << "/" << kIterations;
@@ -592,7 +592,7 @@
const size_t kIterations = 10;
bool has_negotiated = false;
- webrtc::DataChannelInit init;
+ DataChannelInit init;
for (size_t repeats = 0; repeats < kIterations; ++repeats) {
RTC_LOG(LS_INFO) << "Iteration " << (repeats + 1) << "/" << kIterations;
diff --git a/pc/data_channel_unittest.cc b/pc/data_channel_unittest.cc
index 9b84a1b..a27a66c 100644
--- a/pc/data_channel_unittest.cc
+++ b/pc/data_channel_unittest.cc
@@ -81,8 +81,7 @@
controller_(new FakeDataChannelController(&network_thread_)) {
network_thread_.Start();
inner_channel_ = controller_->CreateDataChannel("test", init_);
- channel_ =
- webrtc::SctpDataChannel::CreateProxy(inner_channel_, signaling_safety_);
+ channel_ = SctpDataChannel::CreateProxy(inner_channel_, signaling_safety_);
}
~SctpDataChannelTest() override {
run_loop_.Flush();
@@ -510,7 +509,7 @@
SetChannelReady();
InternalDataChannelInit init;
init.id = 1;
- auto dc = webrtc::SctpDataChannel::CreateProxy(
+ auto dc = SctpDataChannel::CreateProxy(
controller_->CreateDataChannel("test1", init), signaling_safety_);
EXPECT_EQ(DataChannelInterface::kOpen, dc->state());
}
@@ -524,7 +523,7 @@
init.ordered = false;
rtc::scoped_refptr<SctpDataChannel> dc =
controller_->CreateDataChannel("test1", init);
- auto proxy = webrtc::SctpDataChannel::CreateProxy(dc, signaling_safety_);
+ auto proxy = SctpDataChannel::CreateProxy(dc, signaling_safety_);
EXPECT_EQ_WAIT(DataChannelInterface::kOpen, proxy->state(), 1000);
@@ -553,7 +552,7 @@
init.ordered = false;
rtc::scoped_refptr<SctpDataChannel> dc =
controller_->CreateDataChannel("test1", init);
- auto proxy = webrtc::SctpDataChannel::CreateProxy(dc, signaling_safety_);
+ auto proxy = SctpDataChannel::CreateProxy(dc, signaling_safety_);
EXPECT_EQ_WAIT(DataChannelInterface::kOpen, proxy->state(), 1000);
@@ -582,7 +581,7 @@
init.ordered = false;
rtc::scoped_refptr<SctpDataChannel> dc =
controller_->CreateDataChannel("test1", init);
- auto proxy = webrtc::SctpDataChannel::CreateProxy(dc, signaling_safety_);
+ auto proxy = SctpDataChannel::CreateProxy(dc, signaling_safety_);
EXPECT_EQ_WAIT(DataChannelInterface::kOpen, proxy->state(), 1000);
@@ -605,7 +604,7 @@
init.ordered = false;
rtc::scoped_refptr<SctpDataChannel> dc =
controller_->CreateDataChannel("test1", init);
- auto proxy = webrtc::SctpDataChannel::CreateProxy(dc, signaling_safety_);
+ auto proxy = SctpDataChannel::CreateProxy(dc, signaling_safety_);
EXPECT_EQ_WAIT(DataChannelInterface::kOpen, proxy->state(), 1000);
@@ -714,7 +713,7 @@
SetChannelReady();
rtc::scoped_refptr<SctpDataChannel> dc =
controller_->CreateDataChannel("test1", config);
- auto proxy = webrtc::SctpDataChannel::CreateProxy(dc, signaling_safety_);
+ auto proxy = SctpDataChannel::CreateProxy(dc, signaling_safety_);
EXPECT_EQ_WAIT(DataChannelInterface::kOpen, proxy->state(), 1000);
EXPECT_EQ(0, controller_->last_sid());
@@ -779,7 +778,7 @@
SetChannelReady();
rtc::scoped_refptr<SctpDataChannel> dc =
controller_->CreateDataChannel("test1", config);
- auto proxy = webrtc::SctpDataChannel::CreateProxy(dc, signaling_safety_);
+ auto proxy = SctpDataChannel::CreateProxy(dc, signaling_safety_);
EXPECT_EQ_WAIT(DataChannelInterface::kOpen, proxy->state(), 1000);
diff --git a/pc/ice_server_parsing_unittest.cc b/pc/ice_server_parsing_unittest.cc
index 4356b1e..a38638e 100644
--- a/pc/ice_server_parsing_unittest.cc
+++ b/pc/ice_server_parsing_unittest.cc
@@ -62,9 +62,7 @@
server.tls_cert_policy = tls_certificate_policy;
server.hostname = hostname;
servers.push_back(server);
- return webrtc::ParseIceServersOrError(servers, &stun_servers_,
- &turn_servers_)
- .ok();
+ return ParseIceServersOrError(servers, &stun_servers_, &turn_servers_).ok();
}
protected:
@@ -233,8 +231,7 @@
server.password = "bar";
servers.push_back(server);
EXPECT_TRUE(
- webrtc::ParseIceServersOrError(servers, &stun_servers_, &turn_servers_)
- .ok());
+ ParseIceServersOrError(servers, &stun_servers_, &turn_servers_).ok());
EXPECT_EQ(1U, stun_servers_.size());
EXPECT_EQ(1U, turn_servers_.size());
}
diff --git a/pc/ice_transport_unittest.cc b/pc/ice_transport_unittest.cc
index aaf9f2e..a42c107 100644
--- a/pc/ice_transport_unittest.cc
+++ b/pc/ice_transport_unittest.cc
@@ -32,7 +32,7 @@
rtc::SocketServer* socket_server() const { return socket_server_.get(); }
- webrtc::test::ScopedKeyValueConfig field_trials_;
+ test::ScopedKeyValueConfig field_trials_;
private:
std::unique_ptr<rtc::SocketServer> socket_server_;
diff --git a/pc/jsep_transport_controller.cc b/pc/jsep_transport_controller.cc
index 7c669a5..2a701cc 100644
--- a/pc/jsep_transport_controller.cc
+++ b/pc/jsep_transport_controller.cc
@@ -148,7 +148,7 @@
return jsep_transport->rtcp_dtls_transport();
}
-rtc::scoped_refptr<webrtc::DtlsTransport>
+rtc::scoped_refptr<DtlsTransport>
JsepTransportController::LookupDtlsTransportByMid(const std::string& mid) {
RTC_DCHECK_RUN_ON(network_thread_);
auto jsep_transport = GetJsepTransportForMid(mid);
@@ -383,7 +383,7 @@
return RTCError::OK();
}
-rtc::scoped_refptr<webrtc::IceTransportInterface>
+rtc::scoped_refptr<IceTransportInterface>
JsepTransportController::CreateIceTransport(const std::string& transport_name,
bool rtcp) {
int component = rtcp ? cricket::ICE_CANDIDATE_COMPONENT_RTCP
@@ -455,7 +455,7 @@
return dtls;
}
-std::unique_ptr<webrtc::RtpTransport>
+std::unique_ptr<RtpTransport>
JsepTransportController::CreateUnencryptedRtpTransport(
const std::string& transport_name,
rtc::PacketTransportInternal* rtp_packet_transport,
@@ -470,13 +470,12 @@
return unencrypted_rtp_transport;
}
-std::unique_ptr<webrtc::SrtpTransport>
-JsepTransportController::CreateSdesTransport(
+std::unique_ptr<SrtpTransport> JsepTransportController::CreateSdesTransport(
const std::string& transport_name,
cricket::DtlsTransportInternal* rtp_dtls_transport,
cricket::DtlsTransportInternal* rtcp_dtls_transport) {
RTC_DCHECK_RUN_ON(network_thread_);
- auto srtp_transport = std::make_unique<webrtc::SrtpTransport>(
+ auto srtp_transport = std::make_unique<SrtpTransport>(
rtcp_dtls_transport == nullptr, *config_.field_trials);
RTC_DCHECK(rtp_dtls_transport);
srtp_transport->SetRtpPacketTransport(rtp_dtls_transport);
@@ -489,13 +488,13 @@
return srtp_transport;
}
-std::unique_ptr<webrtc::DtlsSrtpTransport>
+std::unique_ptr<DtlsSrtpTransport>
JsepTransportController::CreateDtlsSrtpTransport(
const std::string& transport_name,
cricket::DtlsTransportInternal* rtp_dtls_transport,
cricket::DtlsTransportInternal* rtcp_dtls_transport) {
RTC_DCHECK_RUN_ON(network_thread_);
- auto dtls_srtp_transport = std::make_unique<webrtc::DtlsSrtpTransport>(
+ auto dtls_srtp_transport = std::make_unique<DtlsSrtpTransport>(
rtcp_dtls_transport == nullptr, *config_.field_trials);
if (config_.enable_external_auth) {
dtls_srtp_transport->EnableExternalAuth();
@@ -985,13 +984,12 @@
const cricket::MediaContentDescription* content_desc =
content_info.media_description();
- const webrtc::RtpExtension* send_time_extension =
- webrtc::RtpExtension::FindHeaderExtensionByUri(
- content_desc->rtp_header_extensions(),
- webrtc::RtpExtension::kAbsSendTimeUri,
+ const RtpExtension* send_time_extension =
+ RtpExtension::FindHeaderExtensionByUri(
+ content_desc->rtp_header_extensions(), RtpExtension::kAbsSendTimeUri,
config_.crypto_options.srtp.enable_encrypted_rtp_header_extensions
- ? webrtc::RtpExtension::kPreferEncryptedExtension
- : webrtc::RtpExtension::kDiscardEncryptedExtension);
+ ? RtpExtension::kPreferEncryptedExtension
+ : RtpExtension::kDiscardEncryptedExtension);
return send_time_extension ? send_time_extension->id : -1;
}
@@ -1039,7 +1037,7 @@
"SDES and DTLS-SRTP cannot be enabled at the same time.");
}
- rtc::scoped_refptr<webrtc::IceTransportInterface> ice =
+ rtc::scoped_refptr<IceTransportInterface> ice =
CreateIceTransport(content_info.name, /*rtcp=*/false);
std::unique_ptr<cricket::DtlsTransportInternal> rtp_dtls_transport =
@@ -1050,7 +1048,7 @@
std::unique_ptr<SrtpTransport> sdes_transport;
std::unique_ptr<DtlsSrtpTransport> dtls_srtp_transport;
- rtc::scoped_refptr<webrtc::IceTransportInterface> rtcp_ice;
+ rtc::scoped_refptr<IceTransportInterface> rtcp_ice;
if (config_.rtcp_mux_policy !=
PeerConnectionInterface::kRtcpMuxPolicyRequire &&
content_info.type == cricket::MediaProtocolType::kRtp) {
@@ -1096,7 +1094,7 @@
OnRtcpPacketReceived_n(buffer, packet_time_ms);
});
jsep_transport->rtp_transport()->SetUnDemuxableRtpPacketReceivedHandler(
- [this](webrtc::RtpPacketReceived& packet) {
+ [this](RtpPacketReceived& packet) {
RTC_DCHECK_RUN_ON(network_thread_);
OnUnDemuxableRtpPacketReceived_n(packet);
});
@@ -1421,7 +1419,7 @@
}
void JsepTransportController::OnUnDemuxableRtpPacketReceived_n(
- const webrtc::RtpPacketReceived& packet) {
+ const RtpPacketReceived& packet) {
RTC_DCHECK(config_.un_demuxable_packet_handler);
config_.un_demuxable_packet_handler(packet);
}
diff --git a/pc/jsep_transport_controller.h b/pc/jsep_transport_controller.h
index 5880e34..8f9b9c8 100644
--- a/pc/jsep_transport_controller.h
+++ b/pc/jsep_transport_controller.h
@@ -112,7 +112,7 @@
rtc::SSLProtocolVersion ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
// `crypto_options` is used to determine if created DTLS transports
// negotiate GCM crypto suites or not.
- webrtc::CryptoOptions crypto_options;
+ CryptoOptions crypto_options;
PeerConnectionInterface::BundlePolicy bundle_policy =
PeerConnectionInterface::kBundlePolicyBalanced;
PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy =
@@ -120,7 +120,7 @@
bool disable_encryption = false;
bool enable_external_auth = false;
// Used to inject the ICE/DTLS transports created externally.
- webrtc::IceTransportFactory* ice_transport_factory = nullptr;
+ IceTransportFactory* ice_transport_factory = nullptr;
cricket::DtlsTransportFactory* dtls_transport_factory = nullptr;
Observer* transport_observer = nullptr;
// Must be provided and valid for the lifetime of the
@@ -140,7 +140,7 @@
std::function<void(rtc::SSLHandshakeError)> on_dtls_handshake_error_;
// Field trials.
- const webrtc::FieldTrialsView* field_trials;
+ const FieldTrialsView* field_trials;
};
// The ICE related events are fired on the `network_thread`.
@@ -174,7 +174,7 @@
const cricket::DtlsTransportInternal* GetRtcpDtlsTransport(
const std::string& mid) const;
// Gets the externally sharable version of the DtlsTransport.
- rtc::scoped_refptr<webrtc::DtlsTransport> LookupDtlsTransportByMid(
+ rtc::scoped_refptr<DtlsTransport> LookupDtlsTransportByMid(
const std::string& mid);
rtc::scoped_refptr<SctpTransport> GetSctpTransport(
const std::string& mid) const;
@@ -399,19 +399,19 @@
std::unique_ptr<cricket::DtlsTransportInternal> CreateDtlsTransport(
const cricket::ContentInfo& content_info,
cricket::IceTransportInternal* ice);
- rtc::scoped_refptr<webrtc::IceTransportInterface> CreateIceTransport(
+ rtc::scoped_refptr<IceTransportInterface> CreateIceTransport(
const std::string& transport_name,
bool rtcp);
- std::unique_ptr<webrtc::RtpTransport> CreateUnencryptedRtpTransport(
+ std::unique_ptr<RtpTransport> CreateUnencryptedRtpTransport(
const std::string& transport_name,
rtc::PacketTransportInternal* rtp_packet_transport,
rtc::PacketTransportInternal* rtcp_packet_transport);
- std::unique_ptr<webrtc::SrtpTransport> CreateSdesTransport(
+ std::unique_ptr<SrtpTransport> CreateSdesTransport(
const std::string& transport_name,
cricket::DtlsTransportInternal* rtp_dtls_transport,
cricket::DtlsTransportInternal* rtcp_dtls_transport);
- std::unique_ptr<webrtc::DtlsSrtpTransport> CreateDtlsSrtpTransport(
+ std::unique_ptr<DtlsSrtpTransport> CreateDtlsSrtpTransport(
const std::string& transport_name,
cricket::DtlsTransportInternal* rtp_dtls_transport,
cricket::DtlsTransportInternal* rtcp_dtls_transport);
@@ -453,7 +453,7 @@
void OnRtcpPacketReceived_n(rtc::CopyOnWriteBuffer* packet,
int64_t packet_time_us)
RTC_RUN_ON(network_thread_);
- void OnUnDemuxableRtpPacketReceived_n(const webrtc::RtpPacketReceived& packet)
+ void OnUnDemuxableRtpPacketReceived_n(const RtpPacketReceived& packet)
RTC_RUN_ON(network_thread_);
void OnDtlsHandshakeError(rtc::SSLHandshakeError error);
diff --git a/pc/jsep_transport_controller_unittest.cc b/pc/jsep_transport_controller_unittest.cc
index faa8842..f5e258c 100644
--- a/pc/jsep_transport_controller_unittest.cc
+++ b/pc/jsep_transport_controller_unittest.cc
@@ -56,7 +56,7 @@
namespace webrtc {
-class FakeIceTransportFactory : public webrtc::IceTransportFactory {
+class FakeIceTransportFactory : public IceTransportFactory {
public:
~FakeIceTransportFactory() override = default;
rtc::scoped_refptr<IceTransportInterface> CreateIceTransport(
@@ -72,7 +72,7 @@
public:
std::unique_ptr<cricket::DtlsTransportInternal> CreateDtlsTransport(
cricket::IceTransportInternal* ice,
- const webrtc::CryptoOptions& crypto_options,
+ const CryptoOptions& crypto_options,
rtc::SSLProtocolVersion max_version) override {
return std::make_unique<FakeDtlsTransport>(
static_cast<cricket::FakeIceTransport*>(ice));
@@ -379,7 +379,7 @@
// Transport controller needs to be destroyed first, because it may issue
// callbacks that modify the changed_*_by_mid in the destructor.
std::unique_ptr<JsepTransportController> transport_controller_;
- webrtc::test::ScopedKeyValueConfig field_trials_;
+ test::ScopedKeyValueConfig field_trials_;
};
TEST_F(JsepTransportControllerTest, GetRtpTransport) {
@@ -425,7 +425,7 @@
// and verify that the resulting container is empty.
auto dtls_transport =
transport_controller_->LookupDtlsTransportByMid(kVideoMid1);
- webrtc::DtlsTransport* my_transport =
+ DtlsTransport* my_transport =
static_cast<DtlsTransport*>(dtls_transport.get());
EXPECT_NE(nullptr, my_transport->internal());
transport_controller_.reset();
@@ -899,7 +899,7 @@
transport_controller_->GetDtlsTransport(kAudioMid1));
fake_audio_dtls->fake_ice_transport()->MaybeStartGathering();
fake_audio_dtls->fake_ice_transport()->SetTransportState(
- webrtc::IceTransportState::kChecking,
+ IceTransportState::kChecking,
cricket::IceTransportState::STATE_CONNECTING);
EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionChecking,
ice_connection_state_, kTimeout);
diff --git a/pc/legacy_stats_collector.cc b/pc/legacy_stats_collector.cc
index 3bc65ee..98b7cb9 100644
--- a/pc/legacy_stats_collector.cc
+++ b/pc/legacy_stats_collector.cc
@@ -355,9 +355,8 @@
report->AddInt64(StatsReport::kStatsValueNameInterframeDelayMaxMs,
info.interframe_delay_max_ms);
- report->AddString(
- StatsReport::kStatsValueNameContentType,
- webrtc::videocontenttypehelpers::ToString(info.content_type));
+ report->AddString(StatsReport::kStatsValueNameContentType,
+ videocontenttypehelpers::ToString(info.content_type));
}
void ExtractStats(const cricket::VideoSenderInfo& info,
@@ -398,9 +397,8 @@
for (const auto& i : ints)
report->AddInt(i.name, i.value);
report->AddString(StatsReport::kStatsValueNameMediaType, "video");
- report->AddString(
- StatsReport::kStatsValueNameContentType,
- webrtc::videocontenttypehelpers::ToString(info.content_type));
+ report->AddString(StatsReport::kStatsValueNameContentType,
+ videocontenttypehelpers::ToString(info.content_type));
}
void ExtractStats(const cricket::BandwidthEstimationInfo& info,
@@ -1033,7 +1031,7 @@
if (pc_->signaling_state() == PeerConnectionInterface::kClosed)
return;
- webrtc::Call::Stats call_stats = pc_->GetCallStats();
+ Call::Stats call_stats = pc_->GetCallStats();
cricket::BandwidthEstimationInfo bwe_info;
bwe_info.available_send_bandwidth = call_stats.send_bandwidth_bps;
bwe_info.available_recv_bandwidth = call_stats.recv_bandwidth_bps;
diff --git a/pc/legacy_stats_collector.h b/pc/legacy_stats_collector.h
index e905b39..1c7aad0 100644
--- a/pc/legacy_stats_collector.h
+++ b/pc/legacy_stats_collector.h
@@ -177,9 +177,9 @@
void ExtractMediaInfo(
const std::map<std::string, std::string>& transport_names_by_mid);
void ExtractSenderInfo();
- webrtc::StatsReport* GetReport(const StatsReport::StatsType& type,
- const std::string& id,
- StatsReport::Direction direction);
+ StatsReport* GetReport(const StatsReport::StatsType& type,
+ const std::string& id,
+ StatsReport::Direction direction);
// Helper method to get stats from the local audio tracks.
void UpdateStatsFromExistingLocalAudioTracks(bool has_remote_tracks);
diff --git a/pc/media_stream_unittest.cc b/pc/media_stream_unittest.cc
index f55ea20..d6c79ef 100644
--- a/pc/media_stream_unittest.cc
+++ b/pc/media_stream_unittest.cc
@@ -91,7 +91,7 @@
ASSERT_EQ(1u, stream_->GetAudioTracks().size());
// Verify the video track.
- scoped_refptr<webrtc::MediaStreamTrackInterface> video_track(
+ scoped_refptr<MediaStreamTrackInterface> video_track(
stream_->GetVideoTracks()[0]);
EXPECT_EQ(0, video_track->id().compare(kVideoTrackId));
EXPECT_TRUE(video_track->enabled());
@@ -105,7 +105,7 @@
EXPECT_TRUE(video_track->enabled());
// Verify the audio track.
- scoped_refptr<webrtc::MediaStreamTrackInterface> audio_track(
+ scoped_refptr<MediaStreamTrackInterface> audio_track(
stream_->GetAudioTracks()[0]);
EXPECT_EQ(0, audio_track->id().compare(kAudioTrackId));
EXPECT_TRUE(audio_track->enabled());
@@ -139,14 +139,12 @@
}
TEST_F(MediaStreamTest, ChangeVideoTrack) {
- scoped_refptr<webrtc::VideoTrackInterface> video_track(
- stream_->GetVideoTracks()[0]);
+ scoped_refptr<VideoTrackInterface> video_track(stream_->GetVideoTracks()[0]);
ChangeTrack(video_track.get());
}
TEST_F(MediaStreamTest, ChangeAudioTrack) {
- scoped_refptr<webrtc::AudioTrackInterface> audio_track(
- stream_->GetAudioTracks()[0]);
+ scoped_refptr<AudioTrackInterface> audio_track(stream_->GetAudioTracks()[0]);
ChangeTrack(audio_track.get());
}
diff --git a/pc/peer_connection.cc b/pc/peer_connection.cc
index 183cbeb..46c28bb 100644
--- a/pc/peer_connection.cc
+++ b/pc/peer_connection.cc
@@ -185,7 +185,7 @@
absl::optional<int> RTCConfigurationToIceConfigOptionalInt(
int rtc_configuration_parameter) {
if (rtc_configuration_parameter ==
- webrtc::PeerConnectionInterface::RTCConfiguration::kUndefined) {
+ PeerConnectionInterface::RTCConfiguration::kUndefined) {
return absl::nullopt;
}
return rtc_configuration_parameter;
@@ -449,7 +449,7 @@
absl::optional<int> ice_unwritable_min_checks;
absl::optional<int> ice_inactive_timeout;
absl::optional<int> stun_candidate_keepalive_interval;
- webrtc::TurnCustomizer* turn_customizer;
+ TurnCustomizer* turn_customizer;
SdpSemantics sdp_semantics;
absl::optional<rtc::AdapterType> network_preference;
bool active_reset_srtp_params;
@@ -459,7 +459,7 @@
bool enable_implicit_rollback;
absl::optional<int> report_usage_pattern_delay_ms;
absl::optional<int> stable_writable_connection_ping_interval_ms;
- webrtc::VpnPreference vpn_preference;
+ VpnPreference vpn_preference;
std::vector<rtc::NetworkMask> vpn_list;
PortAllocatorConfig port_allocator_config;
absl::optional<TimeDelta> pacer_burst_interval;
@@ -1685,7 +1685,7 @@
std::function<void(RTCError)> callback) {
RTC_DCHECK_RUN_ON(signaling_thread());
sdp_handler_->AddIceCandidate(std::move(candidate),
- [this, callback](webrtc::RTCError result) {
+ [this, callback](RTCError result) {
ClearStatsCache();
callback(result);
});
@@ -1789,7 +1789,7 @@
std::unique_ptr<RtcEventLogOutput> output) {
int64_t output_period_ms = 5000;
if (trials().IsDisabled("WebRTC-RtcEventLogNewFormat")) {
- output_period_ms = webrtc::RtcEventLog::kImmediateOutput;
+ output_period_ms = RtcEventLog::kImmediateOutput;
}
return StartRtcEventLog(std::move(output), output_period_ms);
}
@@ -2222,7 +2222,7 @@
IceTransportsType type,
int candidate_pool_size,
PortPrunePolicy turn_port_prune_policy,
- webrtc::TurnCustomizer* turn_customizer,
+ TurnCustomizer* turn_customizer,
absl::optional<int> stun_candidate_keepalive_interval,
bool have_local_description) {
RTC_DCHECK_RUN_ON(network_thread());
diff --git a/pc/peer_connection.h b/pc/peer_connection.h
index ea1a9d9..a345089 100644
--- a/pc/peer_connection.h
+++ b/pc/peer_connection.h
@@ -163,7 +163,7 @@
const DataChannelInit* config) override;
// WARNING: LEGACY. See peerconnectioninterface.h
bool GetStats(StatsObserver* observer,
- webrtc::MediaStreamTrackInterface* track,
+ MediaStreamTrackInterface* track,
StatsOutputLevel level) override;
// Spec-complaint GetStats(). See peerconnectioninterface.h
void GetStats(RTCStatsCollectorCallback* callback) override;
@@ -510,7 +510,7 @@
IceTransportsType type,
int candidate_pool_size,
PortPrunePolicy turn_port_prune_policy,
- webrtc::TurnCustomizer* turn_customizer,
+ TurnCustomizer* turn_customizer,
absl::optional<int> stun_candidate_keepalive_interval,
bool have_local_description);
@@ -602,7 +602,7 @@
// a) Specified in PeerConnectionDependencies (owned).
// b) Accessed via ConnectionContext (e.g PeerConnectionFactoryDependencies>
// c) Created as Default (FieldTrialBasedConfig).
- const webrtc::AlwaysValidPointer<const FieldTrialsView, FieldTrialBasedConfig>
+ const AlwaysValidPointer<const FieldTrialsView, FieldTrialBasedConfig>
trials_;
const PeerConnectionFactoryInterface::Options options_;
PeerConnectionObserver* observer_ RTC_GUARDED_BY(signaling_thread()) =
@@ -634,7 +634,7 @@
std::unique_ptr<cricket::PortAllocator>
port_allocator_; // TODO(bugs.webrtc.org/9987): Accessed on both
// signaling and network thread.
- const std::unique_ptr<webrtc::IceTransportFactory>
+ const std::unique_ptr<IceTransportFactory>
ice_transport_factory_; // TODO(bugs.webrtc.org/9987): Accessed on the
// signaling thread but the underlying raw
// pointer is given to
diff --git a/pc/peer_connection_crypto_unittest.cc b/pc/peer_connection_crypto_unittest.cc
index dc350b2..a65988a 100644
--- a/pc/peer_connection_crypto_unittest.cc
+++ b/pc/peer_connection_crypto_unittest.cc
@@ -162,7 +162,7 @@
return transport_info->description.connection_role;
}
- webrtc::test::ScopedKeyValueConfig field_trials_;
+ test::ScopedKeyValueConfig field_trials_;
std::unique_ptr<rtc::VirtualSocketServer> vss_;
rtc::AutoSocketServerThread main_;
rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory_;
diff --git a/pc/peer_connection_encodings_integrationtest.cc b/pc/peer_connection_encodings_integrationtest.cc
index c7181c5..ae23867 100644
--- a/pc/peer_connection_encodings_integrationtest.cc
+++ b/pc/peer_connection_encodings_integrationtest.cc
@@ -77,18 +77,17 @@
// RTX, RED and FEC are reliability mechanisms used in combinations with other
// codecs, but are not themselves a specific codec. Typically you don't want to
// filter these out of the list of codec preferences.
-bool IsReliabilityMechanism(const webrtc::RtpCodecCapability& codec) {
+bool IsReliabilityMechanism(const RtpCodecCapability& codec) {
return absl::EqualsIgnoreCase(codec.name, cricket::kRtxCodecName) ||
absl::EqualsIgnoreCase(codec.name, cricket::kRedCodecName) ||
absl::EqualsIgnoreCase(codec.name, cricket::kUlpfecCodecName);
}
std::string GetCurrentCodecMimeType(
- rtc::scoped_refptr<const webrtc::RTCStatsReport> report,
- const webrtc::RTCOutboundRtpStreamStats& outbound_rtp) {
+ rtc::scoped_refptr<const RTCStatsReport> report,
+ const RTCOutboundRtpStreamStats& outbound_rtp) {
return outbound_rtp.codec_id.is_defined()
- ? *report->GetAs<webrtc::RTCCodecStats>(*outbound_rtp.codec_id)
- ->mime_type
+ ? *report->GetAs<RTCCodecStats>(*outbound_rtp.codec_id)->mime_type
: "";
}
@@ -98,8 +97,8 @@
uint32_t height;
};
-const webrtc::RTCOutboundRtpStreamStats* FindOutboundRtpByRid(
- const std::vector<const webrtc::RTCOutboundRtpStreamStats*>& outbound_rtps,
+const RTCOutboundRtpStreamStats* FindOutboundRtpByRid(
+ const std::vector<const RTCOutboundRtpStreamStats*>& outbound_rtps,
const absl::string_view& rid) {
for (const auto* outbound_rtp : outbound_rtps) {
if (outbound_rtp->rid.is_defined() && *outbound_rtp->rid == rid) {
@@ -121,8 +120,8 @@
rtc::scoped_refptr<PeerConnectionTestWrapper> CreatePc() {
auto pc_wrapper = rtc::make_ref_counted<PeerConnectionTestWrapper>(
"pc", &pss_, background_thread_.get(), background_thread_.get());
- pc_wrapper->CreatePc({}, webrtc::CreateBuiltinAudioEncoderFactory(),
- webrtc::CreateBuiltinAudioDecoderFactory());
+ pc_wrapper->CreatePc({}, CreateBuiltinAudioEncoderFactory(),
+ CreateBuiltinAudioDecoderFactory());
return pc_wrapper;
}
@@ -130,10 +129,9 @@
rtc::scoped_refptr<PeerConnectionTestWrapper> local,
rtc::scoped_refptr<PeerConnectionTestWrapper> remote,
std::vector<cricket::SimulcastLayer> init_layers) {
- rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
- local->GetUserMedia(
- /*audio=*/false, cricket::AudioOptions(), /*video=*/true,
- {.width = 1280, .height = 720});
+ rtc::scoped_refptr<MediaStreamInterface> stream = local->GetUserMedia(
+ /*audio=*/false, cricket::AudioOptions(), /*video=*/true,
+ {.width = 1280, .height = 720});
rtc::scoped_refptr<VideoTrackInterface> track = stream->GetVideoTracks()[0];
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
@@ -973,8 +971,7 @@
local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
rtc::scoped_refptr<RtpTransceiverInterface> audio_transceiver =
transceiver_or_error.MoveValue();
- webrtc::RtpParameters parameters =
- audio_transceiver->sender()->GetParameters();
+ RtpParameters parameters = audio_transceiver->sender()->GetParameters();
EXPECT_FALSE(parameters.encodings[0].codec.has_value());
}
@@ -986,8 +983,7 @@
local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
rtc::scoped_refptr<RtpTransceiverInterface> video_transceiver =
transceiver_or_error.MoveValue();
- webrtc::RtpParameters parameters =
- video_transceiver->sender()->GetParameters();
+ RtpParameters parameters = video_transceiver->sender()->GetParameters();
EXPECT_FALSE(parameters.encodings[0].codec.has_value());
}
@@ -997,19 +993,19 @@
rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc();
ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
- rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
+ rtc::scoped_refptr<MediaStreamInterface> stream =
local_pc_wrapper->GetUserMedia(
/*audio=*/true, {}, /*video=*/false, {});
rtc::scoped_refptr<AudioTrackInterface> track = stream->GetAudioTracks()[0];
- absl::optional<webrtc::RtpCodecCapability> pcmu =
+ absl::optional<RtpCodecCapability> pcmu =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO,
"pcmu");
ASSERT_TRUE(pcmu);
- webrtc::RtpTransceiverInit init;
- init.direction = webrtc::RtpTransceiverDirection::kSendOnly;
- webrtc::RtpEncodingParameters encoding_parameters;
+ RtpTransceiverInit init;
+ init.direction = RtpTransceiverDirection::kSendOnly;
+ RtpEncodingParameters encoding_parameters;
encoding_parameters.codec = pcmu;
init.send_encodings.push_back(encoding_parameters);
@@ -1017,8 +1013,7 @@
local_pc_wrapper->pc()->AddTransceiver(track, init);
rtc::scoped_refptr<RtpTransceiverInterface> audio_transceiver =
transceiver_or_error.MoveValue();
- webrtc::RtpParameters parameters =
- audio_transceiver->sender()->GetParameters();
+ RtpParameters parameters = audio_transceiver->sender()->GetParameters();
EXPECT_EQ(*parameters.encodings[0].codec, *pcmu);
NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper);
@@ -1039,19 +1034,19 @@
rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc();
ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
- rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
+ rtc::scoped_refptr<MediaStreamInterface> stream =
local_pc_wrapper->GetUserMedia(
/*audio=*/false, {}, /*video=*/true, {.width = 1280, .height = 720});
rtc::scoped_refptr<VideoTrackInterface> track = stream->GetVideoTracks()[0];
- absl::optional<webrtc::RtpCodecCapability> vp9 =
+ absl::optional<RtpCodecCapability> vp9 =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
"vp9");
ASSERT_TRUE(vp9);
- webrtc::RtpTransceiverInit init;
- init.direction = webrtc::RtpTransceiverDirection::kSendOnly;
- webrtc::RtpEncodingParameters encoding_parameters;
+ RtpTransceiverInit init;
+ init.direction = RtpTransceiverDirection::kSendOnly;
+ RtpEncodingParameters encoding_parameters;
encoding_parameters.codec = vp9;
encoding_parameters.scalability_mode = "L3T3";
init.send_encodings.push_back(encoding_parameters);
@@ -1060,8 +1055,7 @@
local_pc_wrapper->pc()->AddTransceiver(track, init);
rtc::scoped_refptr<RtpTransceiverInterface> audio_transceiver =
transceiver_or_error.MoveValue();
- webrtc::RtpParameters parameters =
- audio_transceiver->sender()->GetParameters();
+ RtpParameters parameters = audio_transceiver->sender()->GetParameters();
EXPECT_EQ(*parameters.encodings[0].codec, *vp9);
NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper);
@@ -1087,20 +1081,19 @@
rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc();
ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
- rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
+ rtc::scoped_refptr<MediaStreamInterface> stream =
local_pc_wrapper->GetUserMedia(
/*audio=*/true, {}, /*video=*/false, {});
rtc::scoped_refptr<AudioTrackInterface> track = stream->GetAudioTracks()[0];
- absl::optional<webrtc::RtpCodecCapability> pcmu =
+ absl::optional<RtpCodecCapability> pcmu =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO,
"pcmu");
auto transceiver_or_error = local_pc_wrapper->pc()->AddTransceiver(track);
rtc::scoped_refptr<RtpTransceiverInterface> audio_transceiver =
transceiver_or_error.MoveValue();
- webrtc::RtpParameters parameters =
- audio_transceiver->sender()->GetParameters();
+ RtpParameters parameters = audio_transceiver->sender()->GetParameters();
parameters.encodings[0].codec = pcmu;
EXPECT_TRUE(audio_transceiver->sender()->SetParameters(parameters).ok());
@@ -1125,12 +1118,12 @@
rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc();
ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
- rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
+ rtc::scoped_refptr<MediaStreamInterface> stream =
local_pc_wrapper->GetUserMedia(
/*audio=*/true, {}, /*video=*/false, {});
rtc::scoped_refptr<AudioTrackInterface> track = stream->GetAudioTracks()[0];
- absl::optional<webrtc::RtpCodecCapability> pcmu =
+ absl::optional<RtpCodecCapability> pcmu =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO,
"pcmu");
@@ -1150,8 +1143,7 @@
EXPECT_STRCASENE(("audio/" + pcmu->name).c_str(), codec_name.c_str());
std::string last_codec_id = outbound_rtps[0]->codec_id.value();
- webrtc::RtpParameters parameters =
- audio_transceiver->sender()->GetParameters();
+ RtpParameters parameters = audio_transceiver->sender()->GetParameters();
parameters.encodings[0].codec = pcmu;
EXPECT_TRUE(audio_transceiver->sender()->SetParameters(parameters).ok());
@@ -1174,20 +1166,19 @@
rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc();
ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
- rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
+ rtc::scoped_refptr<MediaStreamInterface> stream =
local_pc_wrapper->GetUserMedia(
/*audio=*/false, {}, /*video=*/true, {.width = 1280, .height = 720});
rtc::scoped_refptr<VideoTrackInterface> track = stream->GetVideoTracks()[0];
- absl::optional<webrtc::RtpCodecCapability> vp9 =
+ absl::optional<RtpCodecCapability> vp9 =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
"vp9");
auto transceiver_or_error = local_pc_wrapper->pc()->AddTransceiver(track);
rtc::scoped_refptr<RtpTransceiverInterface> video_transceiver =
transceiver_or_error.MoveValue();
- webrtc::RtpParameters parameters =
- video_transceiver->sender()->GetParameters();
+ RtpParameters parameters = video_transceiver->sender()->GetParameters();
parameters.encodings[0].codec = vp9;
parameters.encodings[0].scalability_mode = "L3T3";
EXPECT_TRUE(video_transceiver->sender()->SetParameters(parameters).ok());
@@ -1218,12 +1209,12 @@
rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc();
ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
- rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
+ rtc::scoped_refptr<MediaStreamInterface> stream =
local_pc_wrapper->GetUserMedia(
/*audio=*/false, {}, /*video=*/true, {.width = 1280, .height = 720});
rtc::scoped_refptr<VideoTrackInterface> track = stream->GetVideoTracks()[0];
- absl::optional<webrtc::RtpCodecCapability> vp9 =
+ absl::optional<RtpCodecCapability> vp9 =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
"vp9");
@@ -1243,8 +1234,7 @@
EXPECT_STRCASENE(("audio/" + vp9->name).c_str(), codec_name.c_str());
std::string last_codec_id = outbound_rtps[0]->codec_id.value();
- webrtc::RtpParameters parameters =
- video_transceiver->sender()->GetParameters();
+ RtpParameters parameters = video_transceiver->sender()->GetParameters();
parameters.encodings[0].codec = vp9;
parameters.encodings[0].scalability_mode = "L3T3";
EXPECT_TRUE(video_transceiver->sender()->SetParameters(parameters).ok());
@@ -1269,15 +1259,15 @@
AddTransceiverRejectsUnknownCodecParameterAudio) {
rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc();
- webrtc::RtpCodec dummy_codec;
+ RtpCodec dummy_codec;
dummy_codec.kind = cricket::MEDIA_TYPE_AUDIO;
dummy_codec.name = "FOOBAR";
dummy_codec.clock_rate = 90000;
dummy_codec.num_channels = 2;
- webrtc::RtpTransceiverInit init;
- init.direction = webrtc::RtpTransceiverDirection::kSendOnly;
- webrtc::RtpEncodingParameters encoding_parameters;
+ RtpTransceiverInit init;
+ init.direction = RtpTransceiverDirection::kSendOnly;
+ RtpEncodingParameters encoding_parameters;
encoding_parameters.codec = dummy_codec;
init.send_encodings.push_back(encoding_parameters);
@@ -1292,14 +1282,14 @@
AddTransceiverRejectsUnknownCodecParameterVideo) {
rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc();
- webrtc::RtpCodec dummy_codec;
+ RtpCodec dummy_codec;
dummy_codec.kind = cricket::MEDIA_TYPE_VIDEO;
dummy_codec.name = "FOOBAR";
dummy_codec.clock_rate = 90000;
- webrtc::RtpTransceiverInit init;
- init.direction = webrtc::RtpTransceiverDirection::kSendOnly;
- webrtc::RtpEncodingParameters encoding_parameters;
+ RtpTransceiverInit init;
+ init.direction = RtpTransceiverDirection::kSendOnly;
+ RtpEncodingParameters encoding_parameters;
encoding_parameters.codec = dummy_codec;
init.send_encodings.push_back(encoding_parameters);
@@ -1314,7 +1304,7 @@
SetParametersRejectsUnknownCodecParameterAudio) {
rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc();
- webrtc::RtpCodec dummy_codec;
+ RtpCodec dummy_codec;
dummy_codec.kind = cricket::MEDIA_TYPE_AUDIO;
dummy_codec.name = "FOOBAR";
dummy_codec.clock_rate = 90000;
@@ -1326,8 +1316,7 @@
rtc::scoped_refptr<RtpTransceiverInterface> audio_transceiver =
transceiver_or_error.MoveValue();
- webrtc::RtpParameters parameters =
- audio_transceiver->sender()->GetParameters();
+ RtpParameters parameters = audio_transceiver->sender()->GetParameters();
parameters.encodings[0].codec = dummy_codec;
RTCError error = audio_transceiver->sender()->SetParameters(parameters);
EXPECT_EQ(error.type(), RTCErrorType::INVALID_MODIFICATION);
@@ -1337,7 +1326,7 @@
SetParametersRejectsUnknownCodecParameterVideo) {
rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc();
- webrtc::RtpCodec dummy_codec;
+ RtpCodec dummy_codec;
dummy_codec.kind = cricket::MEDIA_TYPE_VIDEO;
dummy_codec.name = "FOOBAR";
dummy_codec.clock_rate = 90000;
@@ -1348,8 +1337,7 @@
rtc::scoped_refptr<RtpTransceiverInterface> video_transceiver =
transceiver_or_error.MoveValue();
- webrtc::RtpParameters parameters =
- video_transceiver->sender()->GetParameters();
+ RtpParameters parameters = video_transceiver->sender()->GetParameters();
parameters.encodings[0].codec = dummy_codec;
RTCError error = video_transceiver->sender()->SetParameters(parameters);
EXPECT_EQ(error.type(), RTCErrorType::INVALID_MODIFICATION);
@@ -1359,12 +1347,12 @@
SetParametersRejectsNonPreferredCodecParameterAudio) {
rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc();
- absl::optional<webrtc::RtpCodecCapability> opus =
+ absl::optional<RtpCodecCapability> opus =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO,
"opus");
ASSERT_TRUE(opus);
- std::vector<webrtc::RtpCodecCapability> not_opus_codecs =
+ std::vector<RtpCodecCapability> not_opus_codecs =
local_pc_wrapper->pc_factory()
->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_AUDIO)
.codecs;
@@ -1382,8 +1370,7 @@
transceiver_or_error.MoveValue();
ASSERT_TRUE(audio_transceiver->SetCodecPreferences(not_opus_codecs).ok());
- webrtc::RtpParameters parameters =
- audio_transceiver->sender()->GetParameters();
+ RtpParameters parameters = audio_transceiver->sender()->GetParameters();
parameters.encodings[0].codec = opus;
RTCError error = audio_transceiver->sender()->SetParameters(parameters);
EXPECT_EQ(error.type(), RTCErrorType::INVALID_MODIFICATION);
@@ -1393,12 +1380,12 @@
SetParametersRejectsNonPreferredCodecParameterVideo) {
rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc();
- absl::optional<webrtc::RtpCodecCapability> vp8 =
+ absl::optional<RtpCodecCapability> vp8 =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
"vp8");
ASSERT_TRUE(vp8);
- std::vector<webrtc::RtpCodecCapability> not_vp8_codecs =
+ std::vector<RtpCodecCapability> not_vp8_codecs =
local_pc_wrapper->pc_factory()
->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO)
.codecs;
@@ -1416,8 +1403,7 @@
transceiver_or_error.MoveValue();
ASSERT_TRUE(video_transceiver->SetCodecPreferences(not_vp8_codecs).ok());
- webrtc::RtpParameters parameters =
- video_transceiver->sender()->GetParameters();
+ RtpParameters parameters = video_transceiver->sender()->GetParameters();
parameters.encodings[0].codec = vp8;
RTCError error = video_transceiver->sender()->SetParameters(parameters);
EXPECT_EQ(error.type(), RTCErrorType::INVALID_MODIFICATION);
@@ -1429,12 +1415,12 @@
rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc();
ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
- absl::optional<webrtc::RtpCodecCapability> opus =
+ absl::optional<RtpCodecCapability> opus =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO,
"opus");
ASSERT_TRUE(opus);
- std::vector<webrtc::RtpCodecCapability> not_opus_codecs =
+ std::vector<RtpCodecCapability> not_opus_codecs =
local_pc_wrapper->pc_factory()
->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_AUDIO)
.codecs;
@@ -1456,8 +1442,7 @@
local_pc_wrapper->WaitForConnection();
remote_pc_wrapper->WaitForConnection();
- webrtc::RtpParameters parameters =
- audio_transceiver->sender()->GetParameters();
+ RtpParameters parameters = audio_transceiver->sender()->GetParameters();
parameters.encodings[0].codec = opus;
RTCError error = audio_transceiver->sender()->SetParameters(parameters);
EXPECT_EQ(error.type(), RTCErrorType::INVALID_MODIFICATION);
@@ -1469,12 +1454,12 @@
rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc();
ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
- absl::optional<webrtc::RtpCodecCapability> opus =
+ absl::optional<RtpCodecCapability> opus =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO,
"opus");
ASSERT_TRUE(opus);
- std::vector<webrtc::RtpCodecCapability> not_opus_codecs =
+ std::vector<RtpCodecCapability> not_opus_codecs =
local_pc_wrapper->pc_factory()
->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_AUDIO)
.codecs;
@@ -1519,8 +1504,7 @@
local_pc_wrapper->WaitForConnection();
remote_pc_wrapper->WaitForConnection();
- webrtc::RtpParameters parameters =
- audio_transceiver->sender()->GetParameters();
+ RtpParameters parameters = audio_transceiver->sender()->GetParameters();
parameters.encodings[0].codec = opus;
RTCError error = audio_transceiver->sender()->SetParameters(parameters);
EXPECT_EQ(error.type(), RTCErrorType::INVALID_MODIFICATION);
@@ -1532,12 +1516,12 @@
rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc();
ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
- absl::optional<webrtc::RtpCodecCapability> vp8 =
+ absl::optional<RtpCodecCapability> vp8 =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
"vp8");
ASSERT_TRUE(vp8);
- std::vector<webrtc::RtpCodecCapability> not_vp8_codecs =
+ std::vector<RtpCodecCapability> not_vp8_codecs =
local_pc_wrapper->pc_factory()
->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO)
.codecs;
@@ -1559,8 +1543,7 @@
local_pc_wrapper->WaitForConnection();
remote_pc_wrapper->WaitForConnection();
- webrtc::RtpParameters parameters =
- video_transceiver->sender()->GetParameters();
+ RtpParameters parameters = video_transceiver->sender()->GetParameters();
parameters.encodings[0].codec = vp8;
RTCError error = video_transceiver->sender()->SetParameters(parameters);
EXPECT_EQ(error.type(), RTCErrorType::INVALID_MODIFICATION);
@@ -1572,12 +1555,12 @@
rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc();
ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
- absl::optional<webrtc::RtpCodecCapability> vp8 =
+ absl::optional<RtpCodecCapability> vp8 =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
"vp8");
ASSERT_TRUE(vp8);
- std::vector<webrtc::RtpCodecCapability> not_vp8_codecs =
+ std::vector<RtpCodecCapability> not_vp8_codecs =
local_pc_wrapper->pc_factory()
->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO)
.codecs;
@@ -1622,8 +1605,7 @@
local_pc_wrapper->WaitForConnection();
remote_pc_wrapper->WaitForConnection();
- webrtc::RtpParameters parameters =
- video_transceiver->sender()->GetParameters();
+ RtpParameters parameters = video_transceiver->sender()->GetParameters();
parameters.encodings[0].codec = vp8;
RTCError error = video_transceiver->sender()->SetParameters(parameters);
EXPECT_EQ(error.type(), RTCErrorType::INVALID_MODIFICATION);
@@ -1635,12 +1617,12 @@
rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc();
ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
- absl::optional<webrtc::RtpCodecCapability> opus =
+ absl::optional<RtpCodecCapability> opus =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO,
"opus");
ASSERT_TRUE(opus);
- std::vector<webrtc::RtpCodecCapability> not_opus_codecs =
+ std::vector<RtpCodecCapability> not_opus_codecs =
local_pc_wrapper->pc_factory()
->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_AUDIO)
.codecs;
@@ -1651,9 +1633,9 @@
}),
not_opus_codecs.end());
- webrtc::RtpTransceiverInit init;
- init.direction = webrtc::RtpTransceiverDirection::kSendOnly;
- webrtc::RtpEncodingParameters encoding_parameters;
+ RtpTransceiverInit init;
+ init.direction = RtpTransceiverDirection::kSendOnly;
+ RtpEncodingParameters encoding_parameters;
encoding_parameters.codec = opus;
init.send_encodings.push_back(encoding_parameters);
@@ -1667,8 +1649,7 @@
local_pc_wrapper->WaitForConnection();
remote_pc_wrapper->WaitForConnection();
- webrtc::RtpParameters parameters =
- audio_transceiver->sender()->GetParameters();
+ RtpParameters parameters = audio_transceiver->sender()->GetParameters();
EXPECT_EQ(parameters.encodings[0].codec, opus);
ASSERT_TRUE(audio_transceiver->SetCodecPreferences(not_opus_codecs).ok());
@@ -1684,24 +1665,24 @@
rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc();
ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
- std::vector<webrtc::RtpCodecCapability> send_codecs =
+ std::vector<RtpCodecCapability> send_codecs =
local_pc_wrapper->pc_factory()
->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_AUDIO)
.codecs;
- absl::optional<webrtc::RtpCodecCapability> opus =
+ absl::optional<RtpCodecCapability> opus =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO,
"opus");
ASSERT_TRUE(opus);
- absl::optional<webrtc::RtpCodecCapability> red =
+ absl::optional<RtpCodecCapability> red =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO,
"red");
ASSERT_TRUE(red);
- webrtc::RtpTransceiverInit init;
- init.direction = webrtc::RtpTransceiverDirection::kSendOnly;
- webrtc::RtpEncodingParameters encoding_parameters;
+ RtpTransceiverInit init;
+ init.direction = RtpTransceiverDirection::kSendOnly;
+ RtpEncodingParameters encoding_parameters;
encoding_parameters.codec = opus;
init.send_encodings.push_back(encoding_parameters);
@@ -1720,8 +1701,7 @@
local_pc_wrapper->WaitForConnection();
remote_pc_wrapper->WaitForConnection();
- webrtc::RtpParameters parameters =
- audio_transceiver->sender()->GetParameters();
+ RtpParameters parameters = audio_transceiver->sender()->GetParameters();
EXPECT_EQ(parameters.encodings[0].codec, opus);
EXPECT_EQ(parameters.codecs[0].payload_type, red->preferred_payload_type);
EXPECT_EQ(parameters.codecs[0].name, red->name);
@@ -1743,14 +1723,14 @@
SetParametersRejectsScalabilityModeForSelectedCodec) {
rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc();
- absl::optional<webrtc::RtpCodecCapability> vp8 =
+ absl::optional<RtpCodecCapability> vp8 =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
"vp8");
ASSERT_TRUE(vp8);
- webrtc::RtpTransceiverInit init;
- init.direction = webrtc::RtpTransceiverDirection::kSendOnly;
- webrtc::RtpEncodingParameters encoding_parameters;
+ RtpTransceiverInit init;
+ init.direction = RtpTransceiverDirection::kSendOnly;
+ RtpEncodingParameters encoding_parameters;
encoding_parameters.codec = vp8;
encoding_parameters.scalability_mode = "L1T3";
init.send_encodings.push_back(encoding_parameters);
@@ -1761,8 +1741,7 @@
rtc::scoped_refptr<RtpTransceiverInterface> video_transceiver =
transceiver_or_error.MoveValue();
- webrtc::RtpParameters parameters =
- video_transceiver->sender()->GetParameters();
+ RtpParameters parameters = video_transceiver->sender()->GetParameters();
parameters.encodings[0].scalability_mode = "L3T3";
RTCError error = video_transceiver->sender()->SetParameters(parameters);
EXPECT_EQ(error.type(), RTCErrorType::INVALID_MODIFICATION);
@@ -1774,12 +1753,12 @@
rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc();
ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
- absl::optional<webrtc::RtpCodecCapability> vp8 =
+ absl::optional<RtpCodecCapability> vp8 =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
"vp8");
ASSERT_TRUE(vp8);
- std::vector<webrtc::RtpCodecCapability> not_vp8_codecs =
+ std::vector<RtpCodecCapability> not_vp8_codecs =
local_pc_wrapper->pc_factory()
->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO)
.codecs;
@@ -1790,9 +1769,9 @@
}),
not_vp8_codecs.end());
- webrtc::RtpTransceiverInit init;
- init.direction = webrtc::RtpTransceiverDirection::kSendOnly;
- webrtc::RtpEncodingParameters encoding_parameters;
+ RtpTransceiverInit init;
+ init.direction = RtpTransceiverDirection::kSendOnly;
+ RtpEncodingParameters encoding_parameters;
encoding_parameters.rid = "h";
encoding_parameters.codec = vp8;
encoding_parameters.scale_resolution_down_by = 2;
@@ -1811,8 +1790,7 @@
local_pc_wrapper->WaitForConnection();
remote_pc_wrapper->WaitForConnection();
- webrtc::RtpParameters parameters =
- video_transceiver->sender()->GetParameters();
+ RtpParameters parameters = video_transceiver->sender()->GetParameters();
ASSERT_EQ(parameters.encodings.size(), 2u);
EXPECT_EQ(parameters.encodings[0].codec, vp8);
EXPECT_EQ(parameters.encodings[1].codec, vp8);
@@ -1833,17 +1811,17 @@
rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc();
ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
- absl::optional<webrtc::RtpCodecCapability> vp8 =
+ absl::optional<RtpCodecCapability> vp8 =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
"vp8");
ASSERT_TRUE(vp8);
- absl::optional<webrtc::RtpCodecCapability> vp9 =
+ absl::optional<RtpCodecCapability> vp9 =
local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
"vp9");
- webrtc::RtpTransceiverInit init;
- init.direction = webrtc::RtpTransceiverDirection::kSendOnly;
- webrtc::RtpEncodingParameters encoding_parameters;
+ RtpTransceiverInit init;
+ init.direction = RtpTransceiverDirection::kSendOnly;
+ RtpEncodingParameters encoding_parameters;
encoding_parameters.rid = "h";
encoding_parameters.codec = vp8;
encoding_parameters.scale_resolution_down_by = 2;
diff --git a/pc/peer_connection_factory_proxy.h b/pc/peer_connection_factory_proxy.h
index 4781497..b9bd1cb 100644
--- a/pc/peer_connection_factory_proxy.h
+++ b/pc/peer_connection_factory_proxy.h
@@ -29,10 +29,10 @@
CreatePeerConnectionOrError,
const PeerConnectionInterface::RTCConfiguration&,
PeerConnectionDependencies)
-PROXY_CONSTMETHOD1(webrtc::RtpCapabilities,
+PROXY_CONSTMETHOD1(RtpCapabilities,
GetRtpSenderCapabilities,
cricket::MediaType)
-PROXY_CONSTMETHOD1(webrtc::RtpCapabilities,
+PROXY_CONSTMETHOD1(RtpCapabilities,
GetRtpReceiverCapabilities,
cricket::MediaType)
PROXY_METHOD1(rtc::scoped_refptr<MediaStreamInterface>,
diff --git a/pc/peer_connection_factory_unittest.cc b/pc/peer_connection_factory_unittest.cc
index 91772ec..989b70f 100644
--- a/pc/peer_connection_factory_unittest.cc
+++ b/pc/peer_connection_factory_unittest.cc
@@ -106,8 +106,7 @@
PeerConnectionInterface::IceConnectionState new_state) override {}
void OnIceGatheringChange(
PeerConnectionInterface::IceGatheringState new_state) override {}
- void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override {
- }
+ void OnIceCandidate(const IceCandidateInterface* candidate) override {}
};
class MockNetworkManager : public rtc::NetworkManager {
@@ -133,17 +132,15 @@
private:
void SetUp() {
#ifdef WEBRTC_ANDROID
- webrtc::InitializeAndroidObjects();
+ InitializeAndroidObjects();
#endif
// Use fake audio device module since we're only testing the interface
// level, and using a real one could make tests flaky e.g. when run in
// parallel.
- factory_ = webrtc::CreatePeerConnectionFactory(
+ factory_ = CreatePeerConnectionFactory(
rtc::Thread::Current(), rtc::Thread::Current(), rtc::Thread::Current(),
- rtc::scoped_refptr<webrtc::AudioDeviceModule>(
- FakeAudioCaptureModule::Create()),
- webrtc::CreateBuiltinAudioEncoderFactory(),
- webrtc::CreateBuiltinAudioDecoderFactory(),
+ rtc::scoped_refptr<AudioDeviceModule>(FakeAudioCaptureModule::Create()),
+ CreateBuiltinAudioEncoderFactory(), CreateBuiltinAudioDecoderFactory(),
std::make_unique<VideoEncoderFactoryTemplate<
LibvpxVp8EncoderTemplateAdapter, LibvpxVp9EncoderTemplateAdapter,
OpenH264EncoderTemplateAdapter, LibaomAv1EncoderTemplateAdapter>>(),
@@ -182,64 +179,64 @@
}
}
- void VerifyAudioCodecCapability(const webrtc::RtpCodecCapability& codec) {
+ void VerifyAudioCodecCapability(const RtpCodecCapability& codec) {
EXPECT_EQ(codec.kind, cricket::MEDIA_TYPE_AUDIO);
EXPECT_FALSE(codec.name.empty());
EXPECT_GT(codec.clock_rate, 0);
EXPECT_GT(codec.num_channels, 0);
}
- void VerifyVideoCodecCapability(const webrtc::RtpCodecCapability& codec,
+ void VerifyVideoCodecCapability(const RtpCodecCapability& codec,
bool sender) {
EXPECT_EQ(codec.kind, cricket::MEDIA_TYPE_VIDEO);
EXPECT_FALSE(codec.name.empty());
EXPECT_GT(codec.clock_rate, 0);
if (sender) {
if (codec.name == "VP8" || codec.name == "H264") {
- EXPECT_THAT(codec.scalability_modes,
- UnorderedElementsAre(webrtc::ScalabilityMode::kL1T1,
- webrtc::ScalabilityMode::kL1T2,
- webrtc::ScalabilityMode::kL1T3))
+ EXPECT_THAT(
+ codec.scalability_modes,
+ UnorderedElementsAre(ScalabilityMode::kL1T1, ScalabilityMode::kL1T2,
+ ScalabilityMode::kL1T3))
<< "Codec: " << codec.name;
} else if (codec.name == "VP9" || codec.name == "AV1") {
EXPECT_THAT(
codec.scalability_modes,
UnorderedElementsAre(
// clang-format off
- webrtc::ScalabilityMode::kL1T1,
- webrtc::ScalabilityMode::kL1T2,
- webrtc::ScalabilityMode::kL1T3,
- webrtc::ScalabilityMode::kL2T1,
- webrtc::ScalabilityMode::kL2T1h,
- webrtc::ScalabilityMode::kL2T1_KEY,
- webrtc::ScalabilityMode::kL2T2,
- webrtc::ScalabilityMode::kL2T2h,
- webrtc::ScalabilityMode::kL2T2_KEY,
- webrtc::ScalabilityMode::kL2T2_KEY_SHIFT,
- webrtc::ScalabilityMode::kL2T3,
- webrtc::ScalabilityMode::kL2T3h,
- webrtc::ScalabilityMode::kL2T3_KEY,
- webrtc::ScalabilityMode::kL3T1,
- webrtc::ScalabilityMode::kL3T1h,
- webrtc::ScalabilityMode::kL3T1_KEY,
- webrtc::ScalabilityMode::kL3T2,
- webrtc::ScalabilityMode::kL3T2h,
- webrtc::ScalabilityMode::kL3T2_KEY,
- webrtc::ScalabilityMode::kL3T3,
- webrtc::ScalabilityMode::kL3T3h,
- webrtc::ScalabilityMode::kL3T3_KEY,
- webrtc::ScalabilityMode::kS2T1,
- webrtc::ScalabilityMode::kS2T1h,
- webrtc::ScalabilityMode::kS2T2,
- webrtc::ScalabilityMode::kS2T2h,
- webrtc::ScalabilityMode::kS2T3,
- webrtc::ScalabilityMode::kS2T3h,
- webrtc::ScalabilityMode::kS3T1,
- webrtc::ScalabilityMode::kS3T1h,
- webrtc::ScalabilityMode::kS3T2,
- webrtc::ScalabilityMode::kS3T2h,
- webrtc::ScalabilityMode::kS3T3,
- webrtc::ScalabilityMode::kS3T3h)
+ ScalabilityMode::kL1T1,
+ ScalabilityMode::kL1T2,
+ ScalabilityMode::kL1T3,
+ ScalabilityMode::kL2T1,
+ ScalabilityMode::kL2T1h,
+ ScalabilityMode::kL2T1_KEY,
+ ScalabilityMode::kL2T2,
+ ScalabilityMode::kL2T2h,
+ ScalabilityMode::kL2T2_KEY,
+ ScalabilityMode::kL2T2_KEY_SHIFT,
+ ScalabilityMode::kL2T3,
+ ScalabilityMode::kL2T3h,
+ ScalabilityMode::kL2T3_KEY,
+ ScalabilityMode::kL3T1,
+ ScalabilityMode::kL3T1h,
+ ScalabilityMode::kL3T1_KEY,
+ ScalabilityMode::kL3T2,
+ ScalabilityMode::kL3T2h,
+ ScalabilityMode::kL3T2_KEY,
+ ScalabilityMode::kL3T3,
+ ScalabilityMode::kL3T3h,
+ ScalabilityMode::kL3T3_KEY,
+ ScalabilityMode::kS2T1,
+ ScalabilityMode::kS2T1h,
+ ScalabilityMode::kS2T2,
+ ScalabilityMode::kS2T2h,
+ ScalabilityMode::kS2T3,
+ ScalabilityMode::kS2T3h,
+ ScalabilityMode::kS3T1,
+ ScalabilityMode::kS3T1h,
+ ScalabilityMode::kS3T2,
+ ScalabilityMode::kS3T2h,
+ ScalabilityMode::kS3T3,
+ ScalabilityMode::kS3T3h)
// clang-format on
)
<< "Codec: " << codec.name;
@@ -251,7 +248,7 @@
}
}
- webrtc::test::ScopedKeyValueConfig field_trials_;
+ test::ScopedKeyValueConfig field_trials_;
std::unique_ptr<rtc::SocketServer> socket_server_;
rtc::AutoSocketServerThread main_thread_;
rtc::scoped_refptr<PeerConnectionFactoryInterface> factory_;
@@ -267,7 +264,7 @@
// to reconstruct factory with our own ConnectionContext.
rtc::scoped_refptr<PeerConnectionFactoryInterface>
CreatePeerConnectionFactoryWithRtxDisabled() {
- webrtc::PeerConnectionFactoryDependencies pcf_dependencies;
+ PeerConnectionFactoryDependencies pcf_dependencies;
pcf_dependencies.signaling_thread = rtc::Thread::Current();
pcf_dependencies.worker_thread = rtc::Thread::Current();
pcf_dependencies.network_thread = rtc::Thread::Current();
@@ -287,7 +284,7 @@
OpenH264DecoderTemplateAdapter, Dav1dDecoderTemplateAdapter>>(),
EnableMedia(pcf_dependencies);
- rtc::scoped_refptr<webrtc::ConnectionContext> context =
+ rtc::scoped_refptr<ConnectionContext> context =
ConnectionContext::Create(&pcf_dependencies);
context->set_use_rtx(false);
return rtc::make_ref_counted<PeerConnectionFactory>(context,
@@ -302,26 +299,26 @@
// See https://bugs.chromium.org/p/webrtc/issues/detail?id=7806 for details.
TEST(PeerConnectionFactoryTestInternal, DISABLED_CreatePCUsingInternalModules) {
#ifdef WEBRTC_ANDROID
- webrtc::InitializeAndroidObjects();
+ InitializeAndroidObjects();
#endif
rtc::scoped_refptr<PeerConnectionFactoryInterface> factory(
- webrtc::CreatePeerConnectionFactory(
+ CreatePeerConnectionFactory(
nullptr /* network_thread */, nullptr /* worker_thread */,
nullptr /* signaling_thread */, nullptr /* default_adm */,
- webrtc::CreateBuiltinAudioEncoderFactory(),
- webrtc::CreateBuiltinAudioDecoderFactory(),
+ CreateBuiltinAudioEncoderFactory(),
+ CreateBuiltinAudioDecoderFactory(),
nullptr /* video_encoder_factory */,
nullptr /* video_decoder_factory */, nullptr /* audio_mixer */,
nullptr /* audio_processing */));
NullPeerConnectionObserver observer;
- webrtc::PeerConnectionInterface::RTCConfiguration config;
- config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
+ PeerConnectionInterface::RTCConfiguration config;
+ config.sdp_semantics = SdpSemantics::kUnifiedPlan;
std::unique_ptr<FakeRTCCertificateGenerator> cert_generator(
new FakeRTCCertificateGenerator());
- webrtc::PeerConnectionDependencies pc_dependencies(&observer);
+ PeerConnectionDependencies pc_dependencies(&observer);
pc_dependencies.cert_generator = std::move(cert_generator);
auto result =
factory->CreatePeerConnectionOrError(config, std::move(pc_dependencies));
@@ -330,7 +327,7 @@
}
TEST_F(PeerConnectionFactoryTest, CheckRtpSenderAudioCapabilities) {
- webrtc::RtpCapabilities audio_capabilities =
+ RtpCapabilities audio_capabilities =
factory_->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_AUDIO);
EXPECT_FALSE(audio_capabilities.codecs.empty());
for (const auto& codec : audio_capabilities.codecs) {
@@ -343,7 +340,7 @@
}
TEST_F(PeerConnectionFactoryTest, CheckRtpSenderVideoCapabilities) {
- webrtc::RtpCapabilities video_capabilities =
+ RtpCapabilities video_capabilities =
factory_->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO);
EXPECT_FALSE(video_capabilities.codecs.empty());
for (const auto& codec : video_capabilities.codecs) {
@@ -356,7 +353,7 @@
}
TEST_F(PeerConnectionFactoryTest, CheckRtpSenderRtxEnabledCapabilities) {
- webrtc::RtpCapabilities video_capabilities =
+ RtpCapabilities video_capabilities =
factory_->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO);
const auto it = std::find_if(
video_capabilities.codecs.begin(), video_capabilities.codecs.end(),
@@ -366,7 +363,7 @@
TEST(PeerConnectionFactoryTestInternal, CheckRtpSenderRtxDisabledCapabilities) {
auto factory = CreatePeerConnectionFactoryWithRtxDisabled();
- webrtc::RtpCapabilities video_capabilities =
+ RtpCapabilities video_capabilities =
factory->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO);
const auto it = std::find_if(
video_capabilities.codecs.begin(), video_capabilities.codecs.end(),
@@ -375,14 +372,14 @@
}
TEST_F(PeerConnectionFactoryTest, CheckRtpSenderDataCapabilities) {
- webrtc::RtpCapabilities data_capabilities =
+ RtpCapabilities data_capabilities =
factory_->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_DATA);
EXPECT_TRUE(data_capabilities.codecs.empty());
EXPECT_TRUE(data_capabilities.header_extensions.empty());
}
TEST_F(PeerConnectionFactoryTest, CheckRtpReceiverAudioCapabilities) {
- webrtc::RtpCapabilities audio_capabilities =
+ RtpCapabilities audio_capabilities =
factory_->GetRtpReceiverCapabilities(cricket::MEDIA_TYPE_AUDIO);
EXPECT_FALSE(audio_capabilities.codecs.empty());
for (const auto& codec : audio_capabilities.codecs) {
@@ -395,7 +392,7 @@
}
TEST_F(PeerConnectionFactoryTest, CheckRtpReceiverVideoCapabilities) {
- webrtc::RtpCapabilities video_capabilities =
+ RtpCapabilities video_capabilities =
factory_->GetRtpReceiverCapabilities(cricket::MEDIA_TYPE_VIDEO);
EXPECT_FALSE(video_capabilities.codecs.empty());
for (const auto& codec : video_capabilities.codecs) {
@@ -408,7 +405,7 @@
}
TEST_F(PeerConnectionFactoryTest, CheckRtpReceiverRtxEnabledCapabilities) {
- webrtc::RtpCapabilities video_capabilities =
+ RtpCapabilities video_capabilities =
factory_->GetRtpReceiverCapabilities(cricket::MEDIA_TYPE_VIDEO);
const auto it = std::find_if(
video_capabilities.codecs.begin(), video_capabilities.codecs.end(),
@@ -419,7 +416,7 @@
TEST(PeerConnectionFactoryTestInternal,
CheckRtpReceiverRtxDisabledCapabilities) {
auto factory = CreatePeerConnectionFactoryWithRtxDisabled();
- webrtc::RtpCapabilities video_capabilities =
+ RtpCapabilities video_capabilities =
factory->GetRtpReceiverCapabilities(cricket::MEDIA_TYPE_VIDEO);
const auto it = std::find_if(
video_capabilities.codecs.begin(), video_capabilities.codecs.end(),
@@ -428,7 +425,7 @@
}
TEST_F(PeerConnectionFactoryTest, CheckRtpReceiverDataCapabilities) {
- webrtc::RtpCapabilities data_capabilities =
+ RtpCapabilities data_capabilities =
factory_->GetRtpReceiverCapabilities(cricket::MEDIA_TYPE_DATA);
EXPECT_TRUE(data_capabilities.codecs.empty());
EXPECT_TRUE(data_capabilities.header_extensions.empty());
@@ -438,8 +435,8 @@
// configuration. Also verifies the URL's parsed correctly as expected.
TEST_F(PeerConnectionFactoryTest, CreatePCUsingIceServers) {
PeerConnectionInterface::RTCConfiguration config;
- config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
- webrtc::PeerConnectionInterface::IceServer ice_server;
+ config.sdp_semantics = SdpSemantics::kUnifiedPlan;
+ PeerConnectionInterface::IceServer ice_server;
ice_server.uri = kStunIceServer;
config.servers.push_back(ice_server);
ice_server.uri = kTurnIceServer;
@@ -450,7 +447,7 @@
ice_server.username = kTurnUsername;
ice_server.password = kTurnPassword;
config.servers.push_back(ice_server);
- webrtc::PeerConnectionDependencies pc_dependencies(&observer_);
+ PeerConnectionDependencies pc_dependencies(&observer_);
pc_dependencies.cert_generator =
std::make_unique<FakeRTCCertificateGenerator>();
pc_dependencies.allocator = std::move(port_allocator_);
@@ -475,15 +472,15 @@
// configuration. Also verifies the list of URL's parsed correctly as expected.
TEST_F(PeerConnectionFactoryTest, CreatePCUsingIceServersUrls) {
PeerConnectionInterface::RTCConfiguration config;
- config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
- webrtc::PeerConnectionInterface::IceServer ice_server;
+ config.sdp_semantics = SdpSemantics::kUnifiedPlan;
+ PeerConnectionInterface::IceServer ice_server;
ice_server.urls.push_back(kStunIceServer);
ice_server.urls.push_back(kTurnIceServer);
ice_server.urls.push_back(kTurnIceServerWithTransport);
ice_server.username = kTurnUsername;
ice_server.password = kTurnPassword;
config.servers.push_back(ice_server);
- webrtc::PeerConnectionDependencies pc_dependencies(&observer_);
+ PeerConnectionDependencies pc_dependencies(&observer_);
pc_dependencies.cert_generator =
std::make_unique<FakeRTCCertificateGenerator>();
pc_dependencies.allocator = std::move(port_allocator_);
@@ -506,15 +503,15 @@
TEST_F(PeerConnectionFactoryTest, CreatePCUsingNoUsernameInUri) {
PeerConnectionInterface::RTCConfiguration config;
- config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
- webrtc::PeerConnectionInterface::IceServer ice_server;
+ config.sdp_semantics = SdpSemantics::kUnifiedPlan;
+ PeerConnectionInterface::IceServer ice_server;
ice_server.uri = kStunIceServer;
config.servers.push_back(ice_server);
ice_server.uri = kTurnIceServerWithNoUsernameInUri;
ice_server.username = kTurnUsername;
ice_server.password = kTurnPassword;
config.servers.push_back(ice_server);
- webrtc::PeerConnectionDependencies pc_dependencies(&observer_);
+ PeerConnectionDependencies pc_dependencies(&observer_);
pc_dependencies.cert_generator =
std::make_unique<FakeRTCCertificateGenerator>();
pc_dependencies.allocator = std::move(port_allocator_);
@@ -532,13 +529,13 @@
// has transport parameter in it.
TEST_F(PeerConnectionFactoryTest, CreatePCUsingTurnUrlWithTransportParam) {
PeerConnectionInterface::RTCConfiguration config;
- config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
- webrtc::PeerConnectionInterface::IceServer ice_server;
+ config.sdp_semantics = SdpSemantics::kUnifiedPlan;
+ PeerConnectionInterface::IceServer ice_server;
ice_server.uri = kTurnIceServerWithTransport;
ice_server.username = kTurnUsername;
ice_server.password = kTurnPassword;
config.servers.push_back(ice_server);
- webrtc::PeerConnectionDependencies pc_dependencies(&observer_);
+ PeerConnectionDependencies pc_dependencies(&observer_);
pc_dependencies.cert_generator =
std::make_unique<FakeRTCCertificateGenerator>();
pc_dependencies.allocator = std::move(port_allocator_);
@@ -554,8 +551,8 @@
TEST_F(PeerConnectionFactoryTest, CreatePCUsingSecureTurnUrl) {
PeerConnectionInterface::RTCConfiguration config;
- config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
- webrtc::PeerConnectionInterface::IceServer ice_server;
+ config.sdp_semantics = SdpSemantics::kUnifiedPlan;
+ PeerConnectionInterface::IceServer ice_server;
ice_server.uri = kSecureTurnIceServer;
ice_server.username = kTurnUsername;
ice_server.password = kTurnPassword;
@@ -568,7 +565,7 @@
ice_server.username = kTurnUsername;
ice_server.password = kTurnPassword;
config.servers.push_back(ice_server);
- webrtc::PeerConnectionDependencies pc_dependencies(&observer_);
+ PeerConnectionDependencies pc_dependencies(&observer_);
pc_dependencies.cert_generator =
std::make_unique<FakeRTCCertificateGenerator>();
pc_dependencies.allocator = std::move(port_allocator_);
@@ -593,8 +590,8 @@
TEST_F(PeerConnectionFactoryTest, CreatePCUsingIPLiteralAddress) {
PeerConnectionInterface::RTCConfiguration config;
- config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
- webrtc::PeerConnectionInterface::IceServer ice_server;
+ config.sdp_semantics = SdpSemantics::kUnifiedPlan;
+ PeerConnectionInterface::IceServer ice_server;
ice_server.uri = kStunIceServerWithIPv4Address;
config.servers.push_back(ice_server);
ice_server.uri = kStunIceServerWithIPv4AddressWithoutPort;
@@ -607,7 +604,7 @@
ice_server.username = kTurnUsername;
ice_server.password = kTurnPassword;
config.servers.push_back(ice_server);
- webrtc::PeerConnectionDependencies pc_dependencies(&observer_);
+ PeerConnectionDependencies pc_dependencies(&observer_);
pc_dependencies.cert_generator =
std::make_unique<FakeRTCCertificateGenerator>();
pc_dependencies.allocator = std::move(port_allocator_);
@@ -635,8 +632,8 @@
// This test verifies the captured stream is rendered locally using a
// local video track.
TEST_F(PeerConnectionFactoryTest, LocalRendering) {
- rtc::scoped_refptr<webrtc::FakeVideoTrackSource> source =
- webrtc::FakeVideoTrackSource::Create(/*is_screencast=*/false);
+ rtc::scoped_refptr<FakeVideoTrackSource> source =
+ FakeVideoTrackSource::Create(/*is_screencast=*/false);
cricket::FakeFrameSource frame_source(1280, 720,
rtc::kNumMicrosecsPerSec / 30);
@@ -664,7 +661,7 @@
}
TEST(PeerConnectionFactoryDependenciesTest, UsesNetworkManager) {
- constexpr webrtc::TimeDelta kWaitTimeout = webrtc::TimeDelta::Seconds(10);
+ constexpr TimeDelta kWaitTimeout = TimeDelta::Seconds(10);
auto mock_network_manager = std::make_unique<NiceMock<MockNetworkManager>>();
rtc::Event called;
@@ -672,24 +669,24 @@
.Times(AtLeast(1))
.WillRepeatedly(InvokeWithoutArgs([&] { called.Set(); }));
- webrtc::PeerConnectionFactoryDependencies pcf_dependencies;
+ PeerConnectionFactoryDependencies pcf_dependencies;
pcf_dependencies.network_manager = std::move(mock_network_manager);
- rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pcf =
+ rtc::scoped_refptr<PeerConnectionFactoryInterface> pcf =
CreateModularPeerConnectionFactory(std::move(pcf_dependencies));
PeerConnectionInterface::RTCConfiguration config;
config.ice_candidate_pool_size = 2;
NullPeerConnectionObserver observer;
auto pc = pcf->CreatePeerConnectionOrError(
- config, webrtc::PeerConnectionDependencies(&observer));
+ config, PeerConnectionDependencies(&observer));
ASSERT_TRUE(pc.ok());
called.Wait(kWaitTimeout);
}
TEST(PeerConnectionFactoryDependenciesTest, UsesPacketSocketFactory) {
- constexpr webrtc::TimeDelta kWaitTimeout = webrtc::TimeDelta::Seconds(10);
+ constexpr TimeDelta kWaitTimeout = TimeDelta::Seconds(10);
auto mock_socket_factory =
std::make_unique<NiceMock<rtc::MockPacketSocketFactory>>();
@@ -701,10 +698,10 @@
}))
.WillRepeatedly(Return(nullptr));
- webrtc::PeerConnectionFactoryDependencies pcf_dependencies;
+ PeerConnectionFactoryDependencies pcf_dependencies;
pcf_dependencies.packet_socket_factory = std::move(mock_socket_factory);
- rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pcf =
+ rtc::scoped_refptr<PeerConnectionFactoryInterface> pcf =
CreateModularPeerConnectionFactory(std::move(pcf_dependencies));
// By default, localhost addresses are ignored, which makes tests fail if test
@@ -717,7 +714,7 @@
config.ice_candidate_pool_size = 2;
NullPeerConnectionObserver observer;
auto pc = pcf->CreatePeerConnectionOrError(
- config, webrtc::PeerConnectionDependencies(&observer));
+ config, PeerConnectionDependencies(&observer));
ASSERT_TRUE(pc.ok());
called.Wait(kWaitTimeout);
diff --git a/pc/peer_connection_field_trial_tests.cc b/pc/peer_connection_field_trial_tests.cc
index c009475..4cbe249 100644
--- a/pc/peer_connection_field_trial_tests.cc
+++ b/pc/peer_connection_field_trial_tests.cc
@@ -68,7 +68,7 @@
#ifdef WEBRTC_ANDROID
InitializeAndroidObjects();
#endif
- webrtc::PeerConnectionInterface::IceServer ice_server;
+ PeerConnectionInterface::IceServer ice_server;
ice_server.uri = "stun:stun.l.google.com:19302";
config_.servers.push_back(ice_server);
config_.sdp_semantics = SdpSemantics::kUnifiedPlan;
@@ -108,7 +108,7 @@
std::unique_ptr<rtc::SocketServer> socket_server_;
rtc::AutoSocketServerThread main_thread_;
rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory_ = nullptr;
- webrtc::PeerConnectionInterface::RTCConfiguration config_;
+ PeerConnectionInterface::RTCConfiguration config_;
};
// Tests for the dependency descriptor field trial. The dependency descriptor
@@ -133,7 +133,7 @@
media_description1->rtp_header_extensions();
bool found = absl::c_find_if(rtp_header_extensions1,
- [](const webrtc::RtpExtension& rtp_extension) {
+ [](const RtpExtension& rtp_extension) {
return rtp_extension.uri ==
RtpExtension::kDependencyDescriptorUri;
}) != rtp_header_extensions1.end();
@@ -163,14 +163,14 @@
media_description1->rtp_header_extensions();
bool found1 = absl::c_find_if(rtp_header_extensions1,
- [](const webrtc::RtpExtension& rtp_extension) {
+ [](const RtpExtension& rtp_extension) {
return rtp_extension.uri ==
RtpExtension::kDependencyDescriptorUri;
}) != rtp_header_extensions1.end();
EXPECT_FALSE(found1);
std::set<int> existing_ids;
- for (const webrtc::RtpExtension& rtp_extension : rtp_header_extensions1) {
+ for (const RtpExtension& rtp_extension : rtp_header_extensions1) {
existing_ids.insert(rtp_extension.id);
}
@@ -207,7 +207,7 @@
media_description2->rtp_header_extensions();
bool found2 = absl::c_find_if(rtp_header_extensions2,
- [](const webrtc::RtpExtension& rtp_extension) {
+ [](const RtpExtension& rtp_extension) {
return rtp_extension.uri ==
RtpExtension::kDependencyDescriptorUri;
}) != rtp_header_extensions2.end();
diff --git a/pc/peer_connection_header_extension_unittest.cc b/pc/peer_connection_header_extension_unittest.cc
index dd5d4b0..277979b 100644
--- a/pc/peer_connection_header_extension_unittest.cc
+++ b/pc/peer_connection_header_extension_unittest.cc
@@ -114,7 +114,7 @@
pc_factory, result.MoveValue(), std::move(observer));
}
- webrtc::test::ScopedKeyValueConfig field_trials_;
+ test::ScopedKeyValueConfig field_trials_;
std::unique_ptr<rtc::SocketServer> socket_server_;
rtc::AutoSocketServerThread main_thread_;
std::vector<RtpHeaderExtensionCapability> extensions_;
diff --git a/pc/peer_connection_histogram_unittest.cc b/pc/peer_connection_histogram_unittest.cc
index cadb083..973744c 100644
--- a/pc/peer_connection_histogram_unittest.cc
+++ b/pc/peer_connection_histogram_unittest.cc
@@ -94,7 +94,7 @@
class ObserverForUsageHistogramTest : public MockPeerConnectionObserver {
public:
- void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override;
+ void OnIceCandidate(const IceCandidateInterface* candidate) override;
void OnInterestingUsage(int usage_pattern) override {
interesting_usage_detected_ = usage_pattern;
@@ -157,12 +157,11 @@
return static_cast<ObserverForUsageHistogramTest*>(observer())
->HaveDataChannel();
}
- void BufferIceCandidate(const webrtc::IceCandidateInterface* candidate) {
+ void BufferIceCandidate(const IceCandidateInterface* candidate) {
std::string sdp;
EXPECT_TRUE(candidate->ToString(&sdp));
- std::unique_ptr<webrtc::IceCandidateInterface> candidate_copy(
- CreateIceCandidate(candidate->sdp_mid(), candidate->sdp_mline_index(),
- sdp, nullptr));
+ std::unique_ptr<IceCandidateInterface> candidate_copy(CreateIceCandidate(
+ candidate->sdp_mid(), candidate->sdp_mline_index(), sdp, nullptr));
buffered_candidates_.push_back(std::move(candidate_copy));
}
@@ -213,19 +212,18 @@
return true;
}
- webrtc::PeerConnectionInterface::IceGatheringState ice_gathering_state() {
+ PeerConnectionInterface::IceGatheringState ice_gathering_state() {
return pc()->ice_gathering_state();
}
private:
// Candidates that have been sent but not yet configured
- std::vector<std::unique_ptr<webrtc::IceCandidateInterface>>
- buffered_candidates_;
+ std::vector<std::unique_ptr<IceCandidateInterface>> buffered_candidates_;
};
// Buffers candidates until we add them via AddBufferedIceCandidates.
void ObserverForUsageHistogramTest::OnIceCandidate(
- const webrtc::IceCandidateInterface* candidate) {
+ const IceCandidateInterface* candidate) {
// If target is not set, ignore. This happens in one-ended unit tests.
if (candidate_target_) {
this->candidate_target_->BufferIceCandidate(candidate);
@@ -242,12 +240,12 @@
: vss_(new rtc::VirtualSocketServer()),
socket_factory_(new rtc::BasicPacketSocketFactory(vss_.get())),
main_(vss_.get()) {
- webrtc::metrics::Reset();
+ metrics::Reset();
}
WrapperPtr CreatePeerConnection() {
RTCConfiguration config;
- config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
+ config.sdp_semantics = SdpSemantics::kUnifiedPlan;
return CreatePeerConnection(
config, PeerConnectionFactoryInterface::Options(), nullptr);
}
@@ -259,13 +257,13 @@
WrapperPtr CreatePeerConnectionWithMdns(const RTCConfiguration& config) {
auto resolver_factory =
- std::make_unique<NiceMock<webrtc::MockAsyncDnsResolverFactory>>();
+ std::make_unique<NiceMock<MockAsyncDnsResolverFactory>>();
- webrtc::PeerConnectionDependencies deps(nullptr /* observer_in */);
+ PeerConnectionDependencies deps(nullptr /* observer_in */);
auto fake_network = NewFakeNetwork();
fake_network->set_mdns_responder(
- std::make_unique<webrtc::FakeMdnsResponder>(rtc::Thread::Current()));
+ std::make_unique<FakeMdnsResponder>(rtc::Thread::Current()));
fake_network->AddInterface(NextLocalAddress());
std::unique_ptr<cricket::BasicPortAllocator> port_allocator(
@@ -280,7 +278,7 @@
WrapperPtr CreatePeerConnectionWithImmediateReport() {
RTCConfiguration configuration;
- configuration.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
+ configuration.sdp_semantics = SdpSemantics::kUnifiedPlan;
configuration.report_usage_pattern_delay_ms = 0;
return CreatePeerConnection(
configuration, PeerConnectionFactoryInterface::Options(), nullptr);
@@ -361,7 +359,7 @@
// This works correctly only if there is only one sample value
// that has been counted.
// Returns -1 for "not found".
- return webrtc::metrics::MinSample(kUsagePatternMetric);
+ return metrics::MinSample(kUsagePatternMetric);
}
// The PeerConnection's port allocator is tied to the PeerConnection's
@@ -390,10 +388,10 @@
auto pc = CreatePeerConnectionWithImmediateReport();
int expected_fingerprint = MakeUsageFingerprint({});
- EXPECT_METRIC_EQ_WAIT(1, webrtc::metrics::NumSamples(kUsagePatternMetric),
+ EXPECT_METRIC_EQ_WAIT(1, metrics::NumSamples(kUsagePatternMetric),
kDefaultTimeout);
EXPECT_METRIC_EQ(
- 1, webrtc::metrics::NumEvents(kUsagePatternMetric, expected_fingerprint));
+ 1, metrics::NumEvents(kUsagePatternMetric, expected_fingerprint));
}
#ifndef WEBRTC_ANDROID
@@ -418,11 +416,10 @@
UsageEvent::DIRECT_CONNECTION_SELECTED, UsageEvent::CLOSE_CALLED});
// In this case, we may or may not have PRIVATE_CANDIDATE_COLLECTED,
// depending on the machine configuration.
- EXPECT_METRIC_EQ(2, webrtc::metrics::NumSamples(kUsagePatternMetric));
+ EXPECT_METRIC_EQ(2, metrics::NumSamples(kUsagePatternMetric));
EXPECT_METRIC_TRUE(
- webrtc::metrics::NumEvents(kUsagePatternMetric, expected_fingerprint) ==
- 2 ||
- webrtc::metrics::NumEvents(
+ metrics::NumEvents(kUsagePatternMetric, expected_fingerprint) == 2 ||
+ metrics::NumEvents(
kUsagePatternMetric,
expected_fingerprint |
static_cast<int>(UsageEvent::PRIVATE_CANDIDATE_COLLECTED)) == 2);
@@ -463,11 +460,11 @@
UsageEvent::CANDIDATE_COLLECTED, UsageEvent::ADD_ICE_CANDIDATE_SUCCEEDED,
UsageEvent::REMOTE_MDNS_CANDIDATE_ADDED, UsageEvent::ICE_STATE_CONNECTED,
UsageEvent::REMOTE_CANDIDATE_ADDED, UsageEvent::CLOSE_CALLED});
- EXPECT_METRIC_EQ(2, webrtc::metrics::NumSamples(kUsagePatternMetric));
- EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(kUsagePatternMetric,
- expected_fingerprint_caller));
- EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(kUsagePatternMetric,
- expected_fingerprint_callee));
+ EXPECT_METRIC_EQ(2, metrics::NumSamples(kUsagePatternMetric));
+ EXPECT_METRIC_EQ(
+ 1, metrics::NumEvents(kUsagePatternMetric, expected_fingerprint_caller));
+ EXPECT_METRIC_EQ(
+ 1, metrics::NumEvents(kUsagePatternMetric, expected_fingerprint_callee));
}
// Test getting the usage fingerprint when the callee collects an mDNS
@@ -504,11 +501,11 @@
UsageEvent::ADD_ICE_CANDIDATE_SUCCEEDED, UsageEvent::ICE_STATE_CONNECTED,
UsageEvent::REMOTE_CANDIDATE_ADDED,
UsageEvent::DIRECT_CONNECTION_SELECTED, UsageEvent::CLOSE_CALLED});
- EXPECT_METRIC_EQ(2, webrtc::metrics::NumSamples(kUsagePatternMetric));
- EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(kUsagePatternMetric,
- expected_fingerprint_caller));
- EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(kUsagePatternMetric,
- expected_fingerprint_callee));
+ EXPECT_METRIC_EQ(2, metrics::NumSamples(kUsagePatternMetric));
+ EXPECT_METRIC_EQ(
+ 1, metrics::NumEvents(kUsagePatternMetric, expected_fingerprint_caller));
+ EXPECT_METRIC_EQ(
+ 1, metrics::NumEvents(kUsagePatternMetric, expected_fingerprint_callee));
}
#ifdef WEBRTC_HAVE_SCTP
@@ -526,11 +523,10 @@
UsageEvent::CANDIDATE_COLLECTED, UsageEvent::ADD_ICE_CANDIDATE_SUCCEEDED,
UsageEvent::ICE_STATE_CONNECTED, UsageEvent::REMOTE_CANDIDATE_ADDED,
UsageEvent::DIRECT_CONNECTION_SELECTED, UsageEvent::CLOSE_CALLED});
- EXPECT_METRIC_EQ(2, webrtc::metrics::NumSamples(kUsagePatternMetric));
+ EXPECT_METRIC_EQ(2, metrics::NumSamples(kUsagePatternMetric));
EXPECT_METRIC_TRUE(
- webrtc::metrics::NumEvents(kUsagePatternMetric, expected_fingerprint) ==
- 2 ||
- webrtc::metrics::NumEvents(
+ metrics::NumEvents(kUsagePatternMetric, expected_fingerprint) == 2 ||
+ metrics::NumEvents(
kUsagePatternMetric,
expected_fingerprint |
static_cast<int>(UsageEvent::PRIVATE_CANDIDATE_COLLECTED)) == 2);
@@ -554,9 +550,9 @@
int expected_fingerprint = MakeUsageFingerprint(
{UsageEvent::STUN_SERVER_ADDED, UsageEvent::TURN_SERVER_ADDED,
UsageEvent::CLOSE_CALLED});
- EXPECT_METRIC_EQ(1, webrtc::metrics::NumSamples(kUsagePatternMetric));
+ EXPECT_METRIC_EQ(1, metrics::NumSamples(kUsagePatternMetric));
EXPECT_METRIC_EQ(
- 1, webrtc::metrics::NumEvents(kUsagePatternMetric, expected_fingerprint));
+ 1, metrics::NumEvents(kUsagePatternMetric, expected_fingerprint));
}
TEST_F(PeerConnectionUsageHistogramTest, FingerprintStunTurnInReconfiguration) {
@@ -576,9 +572,9 @@
int expected_fingerprint = MakeUsageFingerprint(
{UsageEvent::STUN_SERVER_ADDED, UsageEvent::TURN_SERVER_ADDED,
UsageEvent::CLOSE_CALLED});
- EXPECT_METRIC_EQ(1, webrtc::metrics::NumSamples(kUsagePatternMetric));
+ EXPECT_METRIC_EQ(1, metrics::NumSamples(kUsagePatternMetric));
EXPECT_METRIC_EQ(
- 1, webrtc::metrics::NumEvents(kUsagePatternMetric, expected_fingerprint));
+ 1, metrics::NumEvents(kUsagePatternMetric, expected_fingerprint));
}
TEST_F(PeerConnectionUsageHistogramTest, FingerprintWithPrivateIPCaller) {
@@ -604,11 +600,11 @@
UsageEvent::REMOTE_PRIVATE_CANDIDATE_ADDED,
UsageEvent::ICE_STATE_CONNECTED, UsageEvent::REMOTE_CANDIDATE_ADDED,
UsageEvent::DIRECT_CONNECTION_SELECTED, UsageEvent::CLOSE_CALLED});
- EXPECT_METRIC_EQ(2, webrtc::metrics::NumSamples(kUsagePatternMetric));
- EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(kUsagePatternMetric,
- expected_fingerprint_caller));
- EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(kUsagePatternMetric,
- expected_fingerprint_callee));
+ EXPECT_METRIC_EQ(2, metrics::NumSamples(kUsagePatternMetric));
+ EXPECT_METRIC_EQ(
+ 1, metrics::NumEvents(kUsagePatternMetric, expected_fingerprint_caller));
+ EXPECT_METRIC_EQ(
+ 1, metrics::NumEvents(kUsagePatternMetric, expected_fingerprint_callee));
}
TEST_F(PeerConnectionUsageHistogramTest, FingerprintWithPrivateIpv6Callee) {
@@ -636,11 +632,11 @@
UsageEvent::ADD_ICE_CANDIDATE_SUCCEEDED,
UsageEvent::REMOTE_CANDIDATE_ADDED, UsageEvent::ICE_STATE_CONNECTED,
UsageEvent::DIRECT_CONNECTION_SELECTED, UsageEvent::CLOSE_CALLED});
- EXPECT_METRIC_EQ(2, webrtc::metrics::NumSamples(kUsagePatternMetric));
- EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(kUsagePatternMetric,
- expected_fingerprint_caller));
- EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(kUsagePatternMetric,
- expected_fingerprint_callee));
+ EXPECT_METRIC_EQ(2, metrics::NumSamples(kUsagePatternMetric));
+ EXPECT_METRIC_EQ(
+ 1, metrics::NumEvents(kUsagePatternMetric, expected_fingerprint_caller));
+ EXPECT_METRIC_EQ(
+ 1, metrics::NumEvents(kUsagePatternMetric, expected_fingerprint_callee));
}
#ifndef WEBRTC_ANDROID
@@ -664,7 +660,7 @@
ASSERT_TRUE(caller->SetLocalDescription(caller->CreateOffer()));
// Wait until the gathering completes so that the session description would
// have contained ICE candidates.
- EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceGatheringComplete,
+ EXPECT_EQ_WAIT(PeerConnectionInterface::kIceGatheringComplete,
caller->ice_gathering_state(), kDefaultTimeout);
EXPECT_TRUE(caller->observer()->candidate_gathered());
// Get the current offer that contains candidates and pass it to the callee.
@@ -713,11 +709,11 @@
UsageEvent::REMOTE_PRIVATE_CANDIDATE_ADDED,
UsageEvent::REMOTE_IPV6_CANDIDATE_ADDED, UsageEvent::ICE_STATE_CONNECTED,
UsageEvent::DIRECT_CONNECTION_SELECTED, UsageEvent::CLOSE_CALLED});
- EXPECT_METRIC_EQ(2, webrtc::metrics::NumSamples(kUsagePatternMetric));
- EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(kUsagePatternMetric,
- expected_fingerprint_caller));
- EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(kUsagePatternMetric,
- expected_fingerprint_callee));
+ EXPECT_METRIC_EQ(2, metrics::NumSamples(kUsagePatternMetric));
+ EXPECT_METRIC_EQ(
+ 1, metrics::NumEvents(kUsagePatternMetric, expected_fingerprint_caller));
+ EXPECT_METRIC_EQ(
+ 1, metrics::NumEvents(kUsagePatternMetric, expected_fingerprint_callee));
}
TEST_F(PeerConnectionUsageHistogramTest, NotableUsageNoted) {
@@ -728,7 +724,7 @@
int expected_fingerprint = MakeUsageFingerprint(
{UsageEvent::DATA_ADDED, UsageEvent::SET_LOCAL_DESCRIPTION_SUCCEEDED,
UsageEvent::CANDIDATE_COLLECTED, UsageEvent::CLOSE_CALLED});
- EXPECT_METRIC_EQ(1, webrtc::metrics::NumSamples(kUsagePatternMetric));
+ EXPECT_METRIC_EQ(1, metrics::NumSamples(kUsagePatternMetric));
EXPECT_METRIC_TRUE(
expected_fingerprint == ObservedFingerprint() ||
(expected_fingerprint |
@@ -745,9 +741,9 @@
int expected_fingerprint = MakeUsageFingerprint(
{UsageEvent::DATA_ADDED, UsageEvent::SET_LOCAL_DESCRIPTION_SUCCEEDED,
UsageEvent::CANDIDATE_COLLECTED});
- EXPECT_METRIC_EQ(0, webrtc::metrics::NumSamples(kUsagePatternMetric));
+ EXPECT_METRIC_EQ(0, metrics::NumSamples(kUsagePatternMetric));
caller->GetInternalPeerConnection()->RequestUsagePatternReportForTesting();
- EXPECT_METRIC_EQ_WAIT(1, webrtc::metrics::NumSamples(kUsagePatternMetric),
+ EXPECT_METRIC_EQ_WAIT(1, metrics::NumSamples(kUsagePatternMetric),
kDefaultTimeout);
EXPECT_METRIC_TRUE(
expected_fingerprint == ObservedFingerprint() ||
@@ -766,12 +762,12 @@
int expected_fingerprint = MakeUsageFingerprint(
{UsageEvent::DATA_ADDED, UsageEvent::SET_LOCAL_DESCRIPTION_SUCCEEDED,
UsageEvent::CANDIDATE_COLLECTED, UsageEvent::CLOSE_CALLED});
- EXPECT_METRIC_EQ(0, webrtc::metrics::NumSamples(kUsagePatternMetric));
+ EXPECT_METRIC_EQ(0, metrics::NumSamples(kUsagePatternMetric));
caller->pc()->Close();
- EXPECT_METRIC_EQ(1, webrtc::metrics::NumSamples(kUsagePatternMetric));
+ EXPECT_METRIC_EQ(1, metrics::NumSamples(kUsagePatternMetric));
caller->GetInternalPeerConnection()->RequestUsagePatternReportForTesting();
caller->observer()->ClearInterestingUsageDetector();
- EXPECT_METRIC_EQ_WAIT(2, webrtc::metrics::NumSamples(kUsagePatternMetric),
+ EXPECT_METRIC_EQ_WAIT(2, metrics::NumSamples(kUsagePatternMetric),
kDefaultTimeout);
EXPECT_METRIC_TRUE(
expected_fingerprint == ObservedFingerprint() ||
diff --git a/pc/peer_connection_ice_unittest.cc b/pc/peer_connection_ice_unittest.cc
index 532583f..492e108 100644
--- a/pc/peer_connection_ice_unittest.cc
+++ b/pc/peer_connection_ice_unittest.cc
@@ -342,7 +342,7 @@
public ::testing::WithParamInterface<SdpSemantics> {
protected:
PeerConnectionIceTest() : PeerConnectionIceBaseTest(GetParam()) {
- webrtc::metrics::Reset();
+ metrics::Reset();
}
};
@@ -514,7 +514,7 @@
EXPECT_FALSE(caller->pc()->AddIceCandidate(jsep_candidate.get()));
EXPECT_METRIC_THAT(
- webrtc::metrics::Samples("WebRTC.PeerConnection.AddIceCandidate"),
+ metrics::Samples("WebRTC.PeerConnection.AddIceCandidate"),
ElementsAre(Pair(kAddIceCandidateFailNoRemoteDescription, 2)));
}
@@ -1457,7 +1457,7 @@
pc_ = result.MoveValue();
}
- webrtc::test::ScopedKeyValueConfig field_trials_;
+ test::ScopedKeyValueConfig field_trials_;
std::unique_ptr<rtc::SocketServer> socket_server_;
rtc::AutoSocketServerThread main_thread_;
rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory_ = nullptr;
diff --git a/pc/peer_connection_integrationtest.cc b/pc/peer_connection_integrationtest.cc
index d76e5e27..1ea16a6 100644
--- a/pc/peer_connection_integrationtest.cc
+++ b/pc/peer_connection_integrationtest.cc
@@ -124,7 +124,7 @@
// Some things use a time of "0" as a special value, so we need to start out
// the fake clock at a nonzero time.
// TODO(deadbeef): Fix this.
- AdvanceTime(webrtc::TimeDelta::Seconds(1));
+ AdvanceTime(TimeDelta::Seconds(1));
}
// Explicit handle.
@@ -324,7 +324,7 @@
ConnectFakeSignaling();
// Add video tracks with 16:9 aspect ratio, size 1280 x 720.
- webrtc::FakePeriodicVideoSource::Config config;
+ FakePeriodicVideoSource::Config config;
config.width = 1280;
config.height = 720;
config.timestamp_offset_ms = rtc::TimeMillis();
@@ -366,7 +366,7 @@
CreateOneDirectionalPeerConnectionWrappers(/*caller_to_callee=*/true));
ConnectFakeSignaling();
// Add one-directional video, from caller to callee.
- rtc::scoped_refptr<webrtc::VideoTrackInterface> caller_track =
+ rtc::scoped_refptr<VideoTrackInterface> caller_track =
caller()->CreateLocalVideoTrack();
caller()->AddTrack(caller_track);
PeerConnectionInterface::RTCOfferAnswerOptions options;
@@ -391,7 +391,7 @@
CreateOneDirectionalPeerConnectionWrappers(/*caller_to_callee=*/false));
ConnectFakeSignaling();
// Add one-directional video, from callee to caller.
- rtc::scoped_refptr<webrtc::VideoTrackInterface> callee_track =
+ rtc::scoped_refptr<VideoTrackInterface> callee_track =
callee()->CreateLocalVideoTrack();
callee()->AddTrack(callee_track);
PeerConnectionInterface::RTCOfferAnswerOptions options;
@@ -414,14 +414,14 @@
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
// Add one-directional video, from caller to callee.
- rtc::scoped_refptr<webrtc::VideoTrackInterface> caller_track =
+ rtc::scoped_refptr<VideoTrackInterface> caller_track =
caller()->CreateLocalVideoTrack();
caller()->AddTrack(caller_track);
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Add receive video.
- rtc::scoped_refptr<webrtc::VideoTrackInterface> callee_track =
+ rtc::scoped_refptr<VideoTrackInterface> callee_track =
callee()->CreateLocalVideoTrack();
callee()->AddTrack(callee_track);
caller()->CreateAndSetAndSignalOffer();
@@ -438,14 +438,14 @@
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
// Add one-directional video, from callee to caller.
- rtc::scoped_refptr<webrtc::VideoTrackInterface> callee_track =
+ rtc::scoped_refptr<VideoTrackInterface> callee_track =
callee()->CreateLocalVideoTrack();
callee()->AddTrack(callee_track);
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Add send video.
- rtc::scoped_refptr<webrtc::VideoTrackInterface> caller_track =
+ rtc::scoped_refptr<VideoTrackInterface> caller_track =
caller()->CreateLocalVideoTrack();
caller()->AddTrack(caller_track);
caller()->CreateAndSetAndSignalOffer();
@@ -462,15 +462,15 @@
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
// Add send video, from caller to callee.
- rtc::scoped_refptr<webrtc::VideoTrackInterface> caller_track =
+ rtc::scoped_refptr<VideoTrackInterface> caller_track =
caller()->CreateLocalVideoTrack();
- rtc::scoped_refptr<webrtc::RtpSenderInterface> caller_sender =
+ rtc::scoped_refptr<RtpSenderInterface> caller_sender =
caller()->AddTrack(caller_track);
// Add receive video, from callee to caller.
- rtc::scoped_refptr<webrtc::VideoTrackInterface> callee_track =
+ rtc::scoped_refptr<VideoTrackInterface> callee_track =
callee()->CreateLocalVideoTrack();
- rtc::scoped_refptr<webrtc::RtpSenderInterface> callee_sender =
+ rtc::scoped_refptr<RtpSenderInterface> callee_sender =
callee()->AddTrack(callee_track);
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
@@ -494,15 +494,15 @@
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
// Add send video, from caller to callee.
- rtc::scoped_refptr<webrtc::VideoTrackInterface> caller_track =
+ rtc::scoped_refptr<VideoTrackInterface> caller_track =
caller()->CreateLocalVideoTrack();
- rtc::scoped_refptr<webrtc::RtpSenderInterface> caller_sender =
+ rtc::scoped_refptr<RtpSenderInterface> caller_sender =
caller()->AddTrack(caller_track);
// Add receive video, from callee to caller.
- rtc::scoped_refptr<webrtc::VideoTrackInterface> callee_track =
+ rtc::scoped_refptr<VideoTrackInterface> callee_track =
callee()->CreateLocalVideoTrack();
- rtc::scoped_refptr<webrtc::RtpSenderInterface> callee_sender =
+ rtc::scoped_refptr<RtpSenderInterface> callee_sender =
callee()->AddTrack(callee_track);
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
@@ -654,9 +654,9 @@
ConnectFakeSignaling();
// Add rotated video tracks.
caller()->AddTrack(
- caller()->CreateLocalVideoTrackWithRotation(webrtc::kVideoRotation_90));
+ caller()->CreateLocalVideoTrackWithRotation(kVideoRotation_90));
callee()->AddTrack(
- callee()->CreateLocalVideoTrackWithRotation(webrtc::kVideoRotation_270));
+ callee()->CreateLocalVideoTrackWithRotation(kVideoRotation_270));
// Wait for video frames to be received by both sides.
caller()->CreateAndSetAndSignalOffer();
@@ -673,8 +673,8 @@
EXPECT_EQ(4.0 / 3, callee()->local_rendered_aspect_ratio());
EXPECT_EQ(4.0 / 3, callee()->rendered_aspect_ratio());
// Ensure that the CVO bits were surfaced to the renderer.
- EXPECT_EQ(webrtc::kVideoRotation_270, caller()->rendered_rotation());
- EXPECT_EQ(webrtc::kVideoRotation_90, callee()->rendered_rotation());
+ EXPECT_EQ(kVideoRotation_270, caller()->rendered_rotation());
+ EXPECT_EQ(kVideoRotation_90, callee()->rendered_rotation());
}
// Test that when the CVO extension isn't supported, video is rotated the
@@ -684,9 +684,9 @@
ConnectFakeSignaling();
// Add rotated video tracks.
caller()->AddTrack(
- caller()->CreateLocalVideoTrackWithRotation(webrtc::kVideoRotation_90));
+ caller()->CreateLocalVideoTrackWithRotation(kVideoRotation_90));
callee()->AddTrack(
- callee()->CreateLocalVideoTrackWithRotation(webrtc::kVideoRotation_270));
+ callee()->CreateLocalVideoTrackWithRotation(kVideoRotation_270));
// Remove the CVO extension from the offered SDP.
callee()->SetReceivedSdpMunger([](cricket::SessionDescription* desc) {
@@ -710,8 +710,8 @@
EXPECT_EQ(3.0 / 4, callee()->local_rendered_aspect_ratio());
EXPECT_EQ(3.0 / 4, callee()->rendered_aspect_ratio());
// Expect that each endpoint is unaware of the rotation of the other endpoint.
- EXPECT_EQ(webrtc::kVideoRotation_0, caller()->rendered_rotation());
- EXPECT_EQ(webrtc::kVideoRotation_0, callee()->rendered_rotation());
+ EXPECT_EQ(kVideoRotation_0, caller()->rendered_rotation());
+ EXPECT_EQ(kVideoRotation_0, callee()->rendered_rotation());
}
// Test that if the answerer rejects the audio m= section, no audio is sent or
@@ -899,9 +899,9 @@
ConnectFakeSignaling();
// Add audio track, do normal offer/answer.
- rtc::scoped_refptr<webrtc::AudioTrackInterface> track =
+ rtc::scoped_refptr<AudioTrackInterface> track =
caller()->CreateLocalAudioTrack();
- rtc::scoped_refptr<webrtc::RtpSenderInterface> sender =
+ rtc::scoped_refptr<RtpSenderInterface> sender =
caller()->pc()->AddTrack(track, {"stream"}).MoveValue();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
@@ -974,7 +974,7 @@
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
// Add one-directional video, from caller to callee.
- rtc::scoped_refptr<webrtc::VideoTrackInterface> track =
+ rtc::scoped_refptr<VideoTrackInterface> track =
caller()->CreateLocalVideoTrack();
RtpTransceiverInit video_transceiver_init;
@@ -988,7 +988,7 @@
// Add receive direction.
video_sender->SetDirectionWithError(RtpTransceiverDirection::kSendRecv);
- rtc::scoped_refptr<webrtc::VideoTrackInterface> callee_track =
+ rtc::scoped_refptr<VideoTrackInterface> callee_track =
callee()->CreateLocalVideoTrack();
callee()->AddTrack(callee_track);
@@ -1348,11 +1348,11 @@
audio_sender_1->track()->id(), video_sender_1->track()->id(),
audio_sender_2->track()->id(), video_sender_2->track()->id()};
- rtc::scoped_refptr<const webrtc::RTCStatsReport> caller_report =
+ rtc::scoped_refptr<const RTCStatsReport> caller_report =
caller()->NewGetStats();
ASSERT_TRUE(caller_report);
auto outbound_stream_stats =
- caller_report->GetStatsOfType<webrtc::RTCOutboundRtpStreamStats>();
+ caller_report->GetStatsOfType<RTCOutboundRtpStreamStats>();
ASSERT_EQ(outbound_stream_stats.size(), 4u);
std::vector<std::string> outbound_track_ids;
for (const auto& stat : outbound_stream_stats) {
@@ -1373,11 +1373,11 @@
}
EXPECT_THAT(outbound_track_ids, UnorderedElementsAreArray(track_ids));
- rtc::scoped_refptr<const webrtc::RTCStatsReport> callee_report =
+ rtc::scoped_refptr<const RTCStatsReport> callee_report =
callee()->NewGetStats();
ASSERT_TRUE(callee_report);
auto inbound_stream_stats =
- callee_report->GetStatsOfType<webrtc::RTCInboundRtpStreamStats>();
+ callee_report->GetStatsOfType<RTCInboundRtpStreamStats>();
ASSERT_EQ(4u, inbound_stream_stats.size());
std::vector<std::string> inbound_track_ids;
for (const auto& stat : inbound_stream_stats) {
@@ -1412,11 +1412,10 @@
// We received a frame, so we should have nonzero "bytes received" stats for
// the unsignaled stream, if stats are working for it.
- rtc::scoped_refptr<const webrtc::RTCStatsReport> report =
- callee()->NewGetStats();
+ rtc::scoped_refptr<const RTCStatsReport> report = callee()->NewGetStats();
ASSERT_NE(nullptr, report);
auto inbound_stream_stats =
- report->GetStatsOfType<webrtc::RTCInboundRtpStreamStats>();
+ report->GetStatsOfType<RTCInboundRtpStreamStats>();
ASSERT_EQ(1U, inbound_stream_stats.size());
ASSERT_TRUE(inbound_stream_stats[0]->bytes_received.is_defined());
ASSERT_GT(*inbound_stream_stats[0]->bytes_received, 0U);
@@ -1459,12 +1458,10 @@
media_expectations.CalleeExpectsSomeVideo(1);
ASSERT_TRUE(ExpectNewFrames(media_expectations));
- rtc::scoped_refptr<const webrtc::RTCStatsReport> report =
- callee()->NewGetStats();
+ rtc::scoped_refptr<const RTCStatsReport> report = callee()->NewGetStats();
ASSERT_NE(nullptr, report);
- auto inbound_rtps =
- report->GetStatsOfType<webrtc::RTCInboundRtpStreamStats>();
+ auto inbound_rtps = report->GetStatsOfType<RTCInboundRtpStreamStats>();
auto index = FindFirstMediaStatsIndexByKind("audio", inbound_rtps);
ASSERT_GE(index, 0);
EXPECT_TRUE(inbound_rtps[index]->audio_level.is_defined());
@@ -1655,18 +1652,18 @@
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
- EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceGatheringComplete,
+ EXPECT_EQ_WAIT(PeerConnectionInterface::kIceGatheringComplete,
caller()->ice_gathering_state(), kMaxWaitForFramesMs);
- EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceGatheringComplete,
+ EXPECT_EQ_WAIT(PeerConnectionInterface::kIceGatheringComplete,
callee()->ice_gathering_state(), kMaxWaitForFramesMs);
// After the best candidate pair is selected and all candidates are signaled,
// the ICE connection state should reach "complete".
// TODO(deadbeef): Currently, the ICE "controlled" agent (the
// answerer/"callee" by default) only reaches "connected". When this is
// fixed, this test should be updated.
- EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
+ EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionCompleted,
caller()->ice_connection_state(), kDefaultTimeout);
- EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
+ EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionConnected,
callee()->ice_connection_state(), kDefaultTimeout);
}
@@ -1679,9 +1676,9 @@
TEST_P(PeerConnectionIntegrationTest,
IceStatesReachCompletionWithRemoteHostname) {
auto caller_resolver_factory =
- std::make_unique<NiceMock<webrtc::MockAsyncDnsResolverFactory>>();
+ std::make_unique<NiceMock<MockAsyncDnsResolverFactory>>();
auto callee_resolver_factory =
- std::make_unique<NiceMock<webrtc::MockAsyncDnsResolverFactory>>();
+ std::make_unique<NiceMock<MockAsyncDnsResolverFactory>>();
auto callee_async_resolver =
std::make_unique<NiceMock<MockAsyncDnsResolver>>();
auto caller_async_resolver =
@@ -1695,12 +1692,12 @@
// P2PTransportChannel.
EXPECT_CALL(*caller_resolver_factory, Create())
.WillOnce(Return(ByMove(std::move(caller_async_resolver))));
- webrtc::PeerConnectionDependencies caller_deps(nullptr);
+ PeerConnectionDependencies caller_deps(nullptr);
caller_deps.async_dns_resolver_factory = std::move(caller_resolver_factory);
EXPECT_CALL(*callee_resolver_factory, Create())
.WillOnce(Return(ByMove(std::move(callee_async_resolver))));
- webrtc::PeerConnectionDependencies callee_deps(nullptr);
+ PeerConnectionDependencies callee_deps(nullptr);
callee_deps.async_dns_resolver_factory = std::move(callee_resolver_factory);
PeerConnectionInterface::RTCConfiguration config;
@@ -1719,9 +1716,9 @@
// Enable hostname candidates with mDNS names.
caller()->SetMdnsResponder(
- std::make_unique<webrtc::FakeMdnsResponder>(network_thread()));
+ std::make_unique<FakeMdnsResponder>(network_thread()));
callee()->SetMdnsResponder(
- std::make_unique<webrtc::FakeMdnsResponder>(network_thread()));
+ std::make_unique<FakeMdnsResponder>(network_thread()));
SetPortAllocatorFlags(kOnlyLocalPorts, kOnlyLocalPorts);
@@ -1730,18 +1727,18 @@
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
- EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
+ EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionCompleted,
caller()->ice_connection_state(), kDefaultTimeout);
- EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
+ EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionConnected,
callee()->ice_connection_state(), kDefaultTimeout);
// Part of reporting the stats will occur on the network thread, so flush it
// before checking NumEvents.
SendTask(network_thread(), [] {});
- EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(
- "WebRTC.PeerConnection.CandidatePairType_UDP",
- webrtc::kIceCandidatePairHostNameHostName));
+ EXPECT_METRIC_EQ(
+ 1, metrics::NumEvents("WebRTC.PeerConnection.CandidatePairType_UDP",
+ kIceCandidatePairHostNameHostName));
DestroyPeerConnections();
}
@@ -1862,9 +1859,9 @@
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
- EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
+ EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionCompleted,
caller()->ice_connection_state(), kDefaultTimeout);
- EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
+ EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionConnected,
callee()->ice_connection_state(), kDefaultTimeout);
// Part of reporting the stats will occur on the network thread, so flush it
@@ -1872,10 +1869,10 @@
SendTask(network_thread(), [] {});
// TODO(bugs.webrtc.org/9456): Fix it.
- const int num_best_ipv4 = webrtc::metrics::NumEvents(
- "WebRTC.PeerConnection.IPMetrics", webrtc::kBestConnections_IPv4);
- const int num_best_ipv6 = webrtc::metrics::NumEvents(
- "WebRTC.PeerConnection.IPMetrics", webrtc::kBestConnections_IPv6);
+ const int num_best_ipv4 = metrics::NumEvents(
+ "WebRTC.PeerConnection.IPMetrics", kBestConnections_IPv4);
+ const int num_best_ipv6 = metrics::NumEvents(
+ "WebRTC.PeerConnection.IPMetrics", kBestConnections_IPv6);
if (TestIPv6()) {
// When IPv6 is enabled, we should prefer an IPv6 connection over an IPv4
// connection.
@@ -1886,12 +1883,12 @@
EXPECT_METRIC_EQ(0, num_best_ipv6);
}
- EXPECT_METRIC_EQ(0, webrtc::metrics::NumEvents(
- "WebRTC.PeerConnection.CandidatePairType_UDP",
- webrtc::kIceCandidatePairHostHost));
- EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(
- "WebRTC.PeerConnection.CandidatePairType_UDP",
- webrtc::kIceCandidatePairHostPublicHostPublic));
+ EXPECT_METRIC_EQ(
+ 0, metrics::NumEvents("WebRTC.PeerConnection.CandidatePairType_UDP",
+ kIceCandidatePairHostHost));
+ EXPECT_METRIC_EQ(
+ 1, metrics::NumEvents("WebRTC.PeerConnection.CandidatePairType_UDP",
+ kIceCandidatePairHostPublicHostPublic));
}
constexpr uint32_t kFlagsIPv4NoStun = cricket::PORTALLOCATOR_DISABLE_TCP |
@@ -1931,17 +1928,17 @@
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
- EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
+ EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionCompleted,
caller()->ice_connection_state(), kMaxWaitForFramesMs);
- EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
+ EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionConnected,
callee()->ice_connection_state(), kMaxWaitForFramesMs);
// To verify that the ICE restart actually occurs, get
// ufrag/password/candidates before and after restart.
// Create an SDP string of the first audio candidate for both clients.
- const webrtc::IceCandidateCollection* audio_candidates_caller =
+ const IceCandidateCollection* audio_candidates_caller =
caller()->pc()->local_description()->candidates(0);
- const webrtc::IceCandidateCollection* audio_candidates_callee =
+ const IceCandidateCollection* audio_candidates_callee =
callee()->pc()->local_description()->candidates(0);
ASSERT_GT(audio_candidates_caller->count(), 0u);
ASSERT_GT(audio_candidates_callee->count(), 0u);
@@ -1964,9 +1961,9 @@
caller()->SetOfferAnswerOptions(IceRestartOfferAnswerOptions());
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
- EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
+ EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionCompleted,
caller()->ice_connection_state(), kMaxWaitForFramesMs);
- EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
+ EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionConnected,
callee()->ice_connection_state(), kMaxWaitForFramesMs);
// Grab the ufrags/candidates again.
@@ -2141,9 +2138,9 @@
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kMaxWaitForActivationMs);
// Wait for ICE to complete, without any tracks being set.
- EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
+ EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionCompleted,
caller()->ice_connection_state(), kMaxWaitForFramesMs);
- EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
+ EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionConnected,
callee()->ice_connection_state(), kMaxWaitForFramesMs);
// Now set the tracks, and expect frames to immediately start flowing.
EXPECT_TRUE(
@@ -2182,9 +2179,9 @@
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kMaxWaitForActivationMs);
// Wait for ICE to complete, without any tracks being set.
- EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
+ EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionCompleted,
caller()->ice_connection_state(), kMaxWaitForFramesMs);
- EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
+ EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionConnected,
callee()->ice_connection_state(), kMaxWaitForFramesMs);
// Now set the tracks, and expect frames to immediately start flowing.
auto callee_audio_sender = callee()->pc()->GetSenders()[0];
@@ -2279,21 +2276,21 @@
});
PeerConnectionInterface::RTCConfiguration client_1_config;
- webrtc::PeerConnectionInterface::IceServer ice_server_1;
+ PeerConnectionInterface::IceServer ice_server_1;
ice_server_1.urls.push_back("turn:88.88.88.0:3478");
ice_server_1.username = "test";
ice_server_1.password = "test";
client_1_config.servers.push_back(ice_server_1);
- client_1_config.type = webrtc::PeerConnectionInterface::kRelay;
+ client_1_config.type = PeerConnectionInterface::kRelay;
client_1_config.presume_writable_when_fully_relayed = true;
PeerConnectionInterface::RTCConfiguration client_2_config;
- webrtc::PeerConnectionInterface::IceServer ice_server_2;
+ PeerConnectionInterface::IceServer ice_server_2;
ice_server_2.urls.push_back("turn:99.99.99.0:3478");
ice_server_2.username = "test";
ice_server_2.password = "test";
client_2_config.servers.push_back(ice_server_2);
- client_2_config.type = webrtc::PeerConnectionInterface::kRelay;
+ client_2_config.type = PeerConnectionInterface::kRelay;
client_2_config.presume_writable_when_fully_relayed = true;
ASSERT_TRUE(
@@ -2326,22 +2323,22 @@
caller()->AddAudioTrack();
// Call getStats, assert there are no candidates.
- rtc::scoped_refptr<const webrtc::RTCStatsReport> first_report =
+ rtc::scoped_refptr<const RTCStatsReport> first_report =
caller()->NewGetStats();
ASSERT_TRUE(first_report);
auto first_candidate_stats =
- first_report->GetStatsOfType<webrtc::RTCLocalIceCandidateStats>();
+ first_report->GetStatsOfType<RTCLocalIceCandidateStats>();
ASSERT_EQ(first_candidate_stats.size(), 0u);
// Create an offer at the caller and set it as remote description on the
// callee.
caller()->CreateAndSetAndSignalOffer();
// Call getStats again, assert there are candidates now.
- rtc::scoped_refptr<const webrtc::RTCStatsReport> second_report =
+ rtc::scoped_refptr<const RTCStatsReport> second_report =
caller()->NewGetStats();
ASSERT_TRUE(second_report);
auto second_candidate_stats =
- second_report->GetStatsOfType<webrtc::RTCLocalIceCandidateStats>();
+ second_report->GetStatsOfType<RTCLocalIceCandidateStats>();
ASSERT_NE(second_candidate_stats.size(), 0u);
// The fake clock ensures that no time has passed so the cache must have been
@@ -2362,17 +2359,17 @@
kDefaultTimeout, FakeClock());
// Call getStats, assert there are no candidates.
- rtc::scoped_refptr<const webrtc::RTCStatsReport> first_report =
+ rtc::scoped_refptr<const RTCStatsReport> first_report =
caller()->NewGetStats();
ASSERT_TRUE(first_report);
auto first_candidate_stats =
- first_report->GetStatsOfType<webrtc::RTCRemoteIceCandidateStats>();
+ first_report->GetStatsOfType<RTCRemoteIceCandidateStats>();
ASSERT_EQ(first_candidate_stats.size(), 0u);
// Add a "fake" candidate.
absl::optional<RTCError> result;
caller()->pc()->AddIceCandidate(
- absl::WrapUnique(webrtc::CreateIceCandidate(
+ absl::WrapUnique(CreateIceCandidate(
"", 0,
"candidate:2214029314 1 udp 2122260223 127.0.0.1 49152 typ host",
nullptr)),
@@ -2381,11 +2378,11 @@
ASSERT_TRUE(result.value().ok());
// Call getStats again, assert there is a remote candidate now.
- rtc::scoped_refptr<const webrtc::RTCStatsReport> second_report =
+ rtc::scoped_refptr<const RTCStatsReport> second_report =
caller()->NewGetStats();
ASSERT_TRUE(second_report);
auto second_candidate_stats =
- second_report->GetStatsOfType<webrtc::RTCRemoteIceCandidateStats>();
+ second_report->GetStatsOfType<RTCRemoteIceCandidateStats>();
ASSERT_EQ(second_candidate_stats.size(), 1u);
// The fake clock ensures that no time has passed so the cache must have been
@@ -2413,22 +2410,22 @@
turn_server_2_external_address);
PeerConnectionInterface::RTCConfiguration client_1_config;
- webrtc::PeerConnectionInterface::IceServer ice_server_1;
+ PeerConnectionInterface::IceServer ice_server_1;
ice_server_1.urls.push_back("turn:88.88.88.0:3478");
ice_server_1.username = "test";
ice_server_1.password = "test";
client_1_config.servers.push_back(ice_server_1);
- client_1_config.type = webrtc::PeerConnectionInterface::kRelay;
+ client_1_config.type = PeerConnectionInterface::kRelay;
auto* customizer1 = CreateTurnCustomizer();
client_1_config.turn_customizer = customizer1;
PeerConnectionInterface::RTCConfiguration client_2_config;
- webrtc::PeerConnectionInterface::IceServer ice_server_2;
+ PeerConnectionInterface::IceServer ice_server_2;
ice_server_2.urls.push_back("turn:99.99.99.0:3478");
ice_server_2.username = "test";
ice_server_2.password = "test";
client_2_config.servers.push_back(ice_server_2);
- client_2_config.type = webrtc::PeerConnectionInterface::kRelay;
+ client_2_config.type = PeerConnectionInterface::kRelay;
auto* customizer2 = CreateTurnCustomizer();
client_2_config.turn_customizer = customizer2;
@@ -2460,18 +2457,18 @@
CreateTurnServer(turn_server_internal_address, turn_server_external_address,
cricket::PROTO_TCP);
- webrtc::PeerConnectionInterface::IceServer ice_server;
+ PeerConnectionInterface::IceServer ice_server;
ice_server.urls.push_back("turn:88.88.88.0:3478?transport=tcp");
ice_server.username = "test";
ice_server.password = "test";
PeerConnectionInterface::RTCConfiguration client_1_config;
client_1_config.servers.push_back(ice_server);
- client_1_config.type = webrtc::PeerConnectionInterface::kRelay;
+ client_1_config.type = PeerConnectionInterface::kRelay;
PeerConnectionInterface::RTCConfiguration client_2_config;
client_2_config.servers.push_back(ice_server);
- client_2_config.type = webrtc::PeerConnectionInterface::kRelay;
+ client_2_config.type = PeerConnectionInterface::kRelay;
ASSERT_TRUE(
CreatePeerConnectionWrappersWithConfig(client_1_config, client_2_config));
@@ -2482,7 +2479,7 @@
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
- EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
+ EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionConnected,
callee()->ice_connection_state(), kMaxWaitForFramesMs);
MediaExpectations media_expectations;
@@ -2506,20 +2503,20 @@
CreateTurnServer(turn_server_internal_address, turn_server_external_address,
cricket::PROTO_TLS, "88.88.88.0");
- webrtc::PeerConnectionInterface::IceServer ice_server;
+ PeerConnectionInterface::IceServer ice_server;
ice_server.urls.push_back("turns:88.88.88.0:3478?transport=tcp");
ice_server.username = "test";
ice_server.password = "test";
PeerConnectionInterface::RTCConfiguration client_1_config;
client_1_config.servers.push_back(ice_server);
- client_1_config.type = webrtc::PeerConnectionInterface::kRelay;
+ client_1_config.type = PeerConnectionInterface::kRelay;
PeerConnectionInterface::RTCConfiguration client_2_config;
client_2_config.servers.push_back(ice_server);
// Setting the type to kRelay forces the connection to go through a TURN
// server.
- client_2_config.type = webrtc::PeerConnectionInterface::kRelay;
+ client_2_config.type = PeerConnectionInterface::kRelay;
// Get a copy to the pointer so we can verify calls later.
rtc::TestCertificateVerifier* client_1_cert_verifier =
@@ -2530,10 +2527,10 @@
client_2_cert_verifier->verify_certificate_ = true;
// Create the dependencies with the test certificate verifier.
- webrtc::PeerConnectionDependencies client_1_deps(nullptr);
+ PeerConnectionDependencies client_1_deps(nullptr);
client_1_deps.tls_cert_verifier =
std::unique_ptr<rtc::TestCertificateVerifier>(client_1_cert_verifier);
- webrtc::PeerConnectionDependencies client_2_deps(nullptr);
+ PeerConnectionDependencies client_2_deps(nullptr);
client_2_deps.tls_cert_verifier =
std::unique_ptr<rtc::TestCertificateVerifier>(client_2_cert_verifier);
@@ -2644,7 +2641,7 @@
ASSERT_GT(receiver->GetParameters().encodings.size(), 0u);
EXPECT_EQ(receiver->GetParameters().encodings[0].ssrc,
sources[0].source_id());
- EXPECT_EQ(webrtc::RtpSourceType::SSRC, sources[0].source_type());
+ EXPECT_EQ(RtpSourceType::SSRC, sources[0].source_type());
}
TEST_P(PeerConnectionIntegrationTest, GetSourcesVideo) {
@@ -2665,7 +2662,7 @@
ASSERT_GT(sources.size(), 0u);
EXPECT_EQ(receiver->GetParameters().encodings[0].ssrc,
sources[0].source_id());
- EXPECT_EQ(webrtc::RtpSourceType::SSRC, sources[0].source_type());
+ EXPECT_EQ(RtpSourceType::SSRC, sources[0].source_type());
}
TEST_P(PeerConnectionIntegrationTest, UnsignaledSsrcGetSourcesAudio) {
@@ -2684,7 +2681,7 @@
})(),
kDefaultTimeout);
ASSERT_GT(sources.size(), 0u);
- EXPECT_EQ(webrtc::RtpSourceType::SSRC, sources[0].source_type());
+ EXPECT_EQ(RtpSourceType::SSRC, sources[0].source_type());
}
TEST_P(PeerConnectionIntegrationTest, UnsignaledSsrcGetSourcesVideo) {
@@ -2703,7 +2700,7 @@
})(),
kDefaultTimeout);
ASSERT_GT(sources.size(), 0u);
- EXPECT_EQ(webrtc::RtpSourceType::SSRC, sources[0].source_type());
+ EXPECT_EQ(RtpSourceType::SSRC, sources[0].source_type());
}
// Similar to the above test, except instead of waiting until GetSources() is
@@ -2728,7 +2725,7 @@
std::vector<RtpSource> sources = receiver->GetSources();
// SSRC history must not be cleared since the reception of the first frame.
ASSERT_GT(sources.size(), 0u);
- EXPECT_EQ(webrtc::RtpSourceType::SSRC, sources[0].source_type());
+ EXPECT_EQ(RtpSourceType::SSRC, sources[0].source_type());
}
TEST_P(PeerConnectionIntegrationTest, UnsignaledSsrcGetParametersAudio) {
@@ -2791,9 +2788,9 @@
ConnectFakeSignaling();
// Add track using stream 1, do offer/answer.
- rtc::scoped_refptr<webrtc::AudioTrackInterface> track =
+ rtc::scoped_refptr<AudioTrackInterface> track =
caller()->CreateLocalAudioTrack();
- rtc::scoped_refptr<webrtc::RtpSenderInterface> sender =
+ rtc::scoped_refptr<RtpSenderInterface> sender =
caller()->AddTrack(track, {"stream_1"});
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
@@ -2825,8 +2822,8 @@
.WillByDefault(::testing::Return(true));
EXPECT_CALL(*output, Write(::testing::A<absl::string_view>()))
.Times(::testing::AtLeast(1));
- EXPECT_TRUE(caller()->pc()->StartRtcEventLog(
- std::move(output), webrtc::RtcEventLog::kImmediateOutput));
+ EXPECT_TRUE(caller()->pc()->StartRtcEventLog(std::move(output),
+ RtcEventLog::kImmediateOutput));
caller()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
@@ -2900,8 +2897,7 @@
double GetAudioEnergyStat(PeerConnectionIntegrationWrapper* pc) {
auto report = pc->NewGetStats();
- auto inbound_rtps =
- report->GetStatsOfType<webrtc::RTCInboundRtpStreamStats>();
+ auto inbound_rtps = report->GetStatsOfType<RTCInboundRtpStreamStats>();
RTC_CHECK(!inbound_rtps.empty());
auto* inbound_rtp = inbound_rtps[0];
if (!inbound_rtp->total_audio_energy.is_defined()) {
@@ -2974,20 +2970,20 @@
ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
ASSERT_NE(nullptr, caller()->event_log_factory());
ASSERT_NE(nullptr, callee()->event_log_factory());
- webrtc::FakeRtcEventLog* caller_event_log =
+ FakeRtcEventLog* caller_event_log =
caller()->event_log_factory()->last_log_created();
- webrtc::FakeRtcEventLog* callee_event_log =
+ FakeRtcEventLog* callee_event_log =
callee()->event_log_factory()->last_log_created();
ASSERT_NE(nullptr, caller_event_log);
ASSERT_NE(nullptr, callee_event_log);
- int caller_ice_config_count = caller_event_log->GetEventCount(
- webrtc::RtcEvent::Type::IceCandidatePairConfig);
- int caller_ice_event_count = caller_event_log->GetEventCount(
- webrtc::RtcEvent::Type::IceCandidatePairEvent);
- int callee_ice_config_count = callee_event_log->GetEventCount(
- webrtc::RtcEvent::Type::IceCandidatePairConfig);
- int callee_ice_event_count = callee_event_log->GetEventCount(
- webrtc::RtcEvent::Type::IceCandidatePairEvent);
+ int caller_ice_config_count =
+ caller_event_log->GetEventCount(RtcEvent::Type::IceCandidatePairConfig);
+ int caller_ice_event_count =
+ caller_event_log->GetEventCount(RtcEvent::Type::IceCandidatePairEvent);
+ int callee_ice_config_count =
+ callee_event_log->GetEventCount(RtcEvent::Type::IceCandidatePairConfig);
+ int callee_ice_event_count =
+ callee_event_log->GetEventCount(RtcEvent::Type::IceCandidatePairEvent);
EXPECT_LT(0, caller_ice_config_count);
EXPECT_LT(0, caller_ice_event_count);
EXPECT_LT(0, callee_ice_config_count);
@@ -3001,20 +2997,20 @@
CreateTurnServer(turn_server_internal_address, turn_server_external_address);
- webrtc::PeerConnectionInterface::IceServer ice_server;
+ PeerConnectionInterface::IceServer ice_server;
ice_server.urls.push_back("turn:88.88.88.0:3478");
ice_server.username = "test";
ice_server.password = "test";
PeerConnectionInterface::RTCConfiguration caller_config;
caller_config.servers.push_back(ice_server);
- caller_config.type = webrtc::PeerConnectionInterface::kRelay;
+ caller_config.type = PeerConnectionInterface::kRelay;
caller_config.continual_gathering_policy = PeerConnection::GATHER_CONTINUALLY;
caller_config.surface_ice_candidates_on_ice_transport_type_changed = true;
PeerConnectionInterface::RTCConfiguration callee_config;
callee_config.servers.push_back(ice_server);
- callee_config.type = webrtc::PeerConnectionInterface::kRelay;
+ callee_config.type = PeerConnectionInterface::kRelay;
callee_config.continual_gathering_policy = PeerConnection::GATHER_CONTINUALLY;
callee_config.surface_ice_candidates_on_ice_transport_type_changed = true;
@@ -3031,9 +3027,9 @@
// kIceGatheringComplete (see
// P2PTransportChannel::OnCandidatesAllocationDone), and consequently not
// kIceConnectionComplete.
- EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
+ EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionConnected,
caller()->ice_connection_state(), kDefaultTimeout);
- EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
+ EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionConnected,
callee()->ice_connection_state(), kDefaultTimeout);
// Note that we cannot use the metric
// `WebRTC.PeerConnection.CandidatePairType_UDP` in this test since this
@@ -3046,7 +3042,7 @@
// Loosen the caller's candidate filter.
caller_config = caller()->pc()->GetConfiguration();
- caller_config.type = webrtc::PeerConnectionInterface::kAll;
+ caller_config.type = PeerConnectionInterface::kAll;
caller()->pc()->SetConfiguration(caller_config);
// We should have gathered a new host candidate.
EXPECT_EQ_WAIT(cricket::LOCAL_PORT_TYPE,
@@ -3054,7 +3050,7 @@
// Loosen the callee's candidate filter.
callee_config = callee()->pc()->GetConfiguration();
- callee_config.type = webrtc::PeerConnectionInterface::kAll;
+ callee_config.type = PeerConnectionInterface::kAll;
callee()->pc()->SetConfiguration(callee_config);
EXPECT_EQ_WAIT(cricket::LOCAL_PORT_TYPE,
callee()->last_candidate_gathered().type(), kDefaultTimeout);
@@ -3084,19 +3080,19 @@
CreateTurnServer(turn_server_internal_address, turn_server_external_address);
- webrtc::PeerConnectionInterface::IceServer ice_server;
+ PeerConnectionInterface::IceServer ice_server;
ice_server.urls.push_back("turn:88.88.88.0:3478");
ice_server.username = "test";
ice_server.password = "123";
PeerConnectionInterface::RTCConfiguration caller_config;
caller_config.servers.push_back(ice_server);
- caller_config.type = webrtc::PeerConnectionInterface::kRelay;
+ caller_config.type = PeerConnectionInterface::kRelay;
caller_config.continual_gathering_policy = PeerConnection::GATHER_CONTINUALLY;
PeerConnectionInterface::RTCConfiguration callee_config;
callee_config.servers.push_back(ice_server);
- callee_config.type = webrtc::PeerConnectionInterface::kRelay;
+ callee_config.type = PeerConnectionInterface::kRelay;
callee_config.continual_gathering_policy = PeerConnection::GATHER_CONTINUALLY;
ASSERT_TRUE(
@@ -3115,19 +3111,19 @@
}
TEST_P(PeerConnectionIntegrationTest, OnIceCandidateErrorWithEmptyAddress) {
- webrtc::PeerConnectionInterface::IceServer ice_server;
+ PeerConnectionInterface::IceServer ice_server;
ice_server.urls.push_back("turn:127.0.0.1:3478?transport=tcp");
ice_server.username = "test";
ice_server.password = "test";
PeerConnectionInterface::RTCConfiguration caller_config;
caller_config.servers.push_back(ice_server);
- caller_config.type = webrtc::PeerConnectionInterface::kRelay;
+ caller_config.type = PeerConnectionInterface::kRelay;
caller_config.continual_gathering_policy = PeerConnection::GATHER_CONTINUALLY;
PeerConnectionInterface::RTCConfiguration callee_config;
callee_config.servers.push_back(ice_server);
- callee_config.type = webrtc::PeerConnectionInterface::kRelay;
+ callee_config.type = PeerConnectionInterface::kRelay;
callee_config.continual_gathering_policy = PeerConnection::GATHER_CONTINUALLY;
ASSERT_TRUE(
@@ -3697,7 +3693,7 @@
CreateOneDirectionalPeerConnectionWrappers(/*caller_to_callee=*/true));
ConnectFakeSignaling();
// Add one-directional video, from caller to callee.
- rtc::scoped_refptr<webrtc::VideoTrackInterface> caller_track =
+ rtc::scoped_refptr<VideoTrackInterface> caller_track =
caller()->CreateLocalVideoTrack();
auto sender = caller()->AddTrack(caller_track);
PeerConnectionInterface::RTCOfferAnswerOptions options;
@@ -3722,7 +3718,7 @@
}
int NacksReceivedCount(PeerConnectionIntegrationWrapper& pc) {
- rtc::scoped_refptr<const webrtc::RTCStatsReport> report = pc.NewGetStats();
+ rtc::scoped_refptr<const RTCStatsReport> report = pc.NewGetStats();
auto sender_stats = report->GetStatsOfType<RTCOutboundRtpStreamStats>();
if (sender_stats.size() != 1) {
ADD_FAILURE();
@@ -3735,7 +3731,7 @@
}
int NacksSentCount(PeerConnectionIntegrationWrapper& pc) {
- rtc::scoped_refptr<const webrtc::RTCStatsReport> report = pc.NewGetStats();
+ rtc::scoped_refptr<const RTCStatsReport> report = pc.NewGetStats();
auto receiver_stats = report->GetStatsOfType<RTCInboundRtpStreamStats>();
if (receiver_stats.size() != 1) {
ADD_FAILURE();
diff --git a/pc/peer_connection_interface_unittest.cc b/pc/peer_connection_interface_unittest.cc
index 2dbc6f5..c057e55 100644
--- a/pc/peer_connection_interface_unittest.cc
+++ b/pc/peer_connection_interface_unittest.cc
@@ -474,8 +474,7 @@
// Get the ufrags out of an SDP blob. Useful for testing ICE restart
// behavior.
-std::vector<std::string> GetUfrags(
- const webrtc::SessionDescriptionInterface* desc) {
+std::vector<std::string> GetUfrags(const SessionDescriptionInterface* desc) {
std::vector<std::string> ufrags;
for (const cricket::TransportInfo& info :
desc->description()->transport_infos()) {
@@ -544,21 +543,19 @@
StreamCollection::Create());
for (int i = 0; i < number_of_streams; ++i) {
- rtc::scoped_refptr<webrtc::MediaStreamInterface> stream(
- webrtc::MediaStream::Create(kStreams[i]));
+ rtc::scoped_refptr<MediaStreamInterface> stream(
+ MediaStream::Create(kStreams[i]));
for (int j = 0; j < tracks_per_stream; ++j) {
// Add a local audio track.
- rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
- webrtc::AudioTrack::Create(kAudioTracks[i * tracks_per_stream + j],
- nullptr));
+ rtc::scoped_refptr<AudioTrackInterface> audio_track(
+ AudioTrack::Create(kAudioTracks[i * tracks_per_stream + j], nullptr));
stream->AddTrack(audio_track);
// 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()));
+ rtc::scoped_refptr<VideoTrackInterface> video_track(VideoTrack::Create(
+ kVideoTracks[i * tracks_per_stream + j],
+ FakeVideoTrackSource::Create(), rtc::Thread::Current()));
stream->AddTrack(video_track);
}
@@ -578,10 +575,10 @@
if (s1->at(i)->id() != s2->at(i)->id()) {
return false;
}
- webrtc::AudioTrackVector audio_tracks1 = s1->at(i)->GetAudioTracks();
- webrtc::AudioTrackVector audio_tracks2 = s2->at(i)->GetAudioTracks();
- webrtc::VideoTrackVector video_tracks1 = s1->at(i)->GetVideoTracks();
- webrtc::VideoTrackVector video_tracks2 = s2->at(i)->GetVideoTracks();
+ AudioTrackVector audio_tracks1 = s1->at(i)->GetAudioTracks();
+ AudioTrackVector audio_tracks2 = s2->at(i)->GetAudioTracks();
+ VideoTrackVector video_tracks1 = s1->at(i)->GetVideoTracks();
+ VideoTrackVector video_tracks2 = s2->at(i)->GetVideoTracks();
if (audio_tracks1.size() != audio_tracks2.size()) {
return false;
@@ -630,7 +627,7 @@
// constraints are propagated into the PeerConnection's MediaConfig. These
// settings are intended for MediaChannel constructors, but that is not
// exercised by these unittest.
-class PeerConnectionFactoryForTest : public webrtc::PeerConnectionFactory {
+class PeerConnectionFactoryForTest : public PeerConnectionFactory {
public:
static rtc::scoped_refptr<PeerConnectionFactoryForTest>
CreatePeerConnectionFactoryForTest() {
@@ -665,7 +662,7 @@
main_(vss_.get()),
sdp_semantics_(sdp_semantics) {
#ifdef WEBRTC_ANDROID
- webrtc::InitializeAndroidObjects();
+ InitializeAndroidObjects();
#endif
}
@@ -673,22 +670,16 @@
// Use fake audio capture module since we're only testing the interface
// level, and using a real one could make tests flaky when run in parallel.
fake_audio_capture_module_ = FakeAudioCaptureModule::Create();
- pc_factory_ = webrtc::CreatePeerConnectionFactory(
+ pc_factory_ = CreatePeerConnectionFactory(
rtc::Thread::Current(), rtc::Thread::Current(), rtc::Thread::Current(),
- rtc::scoped_refptr<webrtc::AudioDeviceModule>(
- fake_audio_capture_module_),
- webrtc::CreateBuiltinAudioEncoderFactory(),
- webrtc::CreateBuiltinAudioDecoderFactory(),
- std::make_unique<webrtc::VideoEncoderFactoryTemplate<
- webrtc::LibvpxVp8EncoderTemplateAdapter,
- webrtc::LibvpxVp9EncoderTemplateAdapter,
- webrtc::OpenH264EncoderTemplateAdapter,
- webrtc::LibaomAv1EncoderTemplateAdapter>>(),
- std::make_unique<webrtc::VideoDecoderFactoryTemplate<
- webrtc::LibvpxVp8DecoderTemplateAdapter,
- webrtc::LibvpxVp9DecoderTemplateAdapter,
- webrtc::OpenH264DecoderTemplateAdapter,
- webrtc::Dav1dDecoderTemplateAdapter>>(),
+ rtc::scoped_refptr<AudioDeviceModule>(fake_audio_capture_module_),
+ CreateBuiltinAudioEncoderFactory(), CreateBuiltinAudioDecoderFactory(),
+ std::make_unique<VideoEncoderFactoryTemplate<
+ LibvpxVp8EncoderTemplateAdapter, LibvpxVp9EncoderTemplateAdapter,
+ OpenH264EncoderTemplateAdapter, LibaomAv1EncoderTemplateAdapter>>(),
+ std::make_unique<VideoDecoderFactoryTemplate<
+ LibvpxVp8DecoderTemplateAdapter, LibvpxVp9DecoderTemplateAdapter,
+ OpenH264DecoderTemplateAdapter, Dav1dDecoderTemplateAdapter>>(),
nullptr /* audio_mixer */, nullptr /* audio_processing */);
ASSERT_TRUE(pc_factory_);
}
@@ -946,8 +937,7 @@
// Call the standards-compliant GetStats function.
bool DoGetRTCStats() {
- auto callback =
- rtc::make_ref_counted<webrtc::MockRTCStatsCollectorCallback>();
+ auto callback = rtc::make_ref_counted<MockRTCStatsCollectorCallback>();
pc_->GetStats(callback.get());
EXPECT_TRUE_WAIT(callback->called(), kTimeout);
return callback->called();
@@ -987,14 +977,14 @@
std::string sdp;
EXPECT_TRUE(offer->ToString(&sdp));
std::unique_ptr<SessionDescriptionInterface> remote_offer(
- webrtc::CreateSessionDescription(SdpType::kOffer, sdp));
+ CreateSessionDescription(SdpType::kOffer, sdp));
EXPECT_TRUE(DoSetRemoteDescription(std::move(remote_offer)));
EXPECT_EQ(PeerConnectionInterface::kHaveRemoteOffer, observer_.state_);
}
void CreateAndSetRemoteOffer(const std::string& sdp) {
std::unique_ptr<SessionDescriptionInterface> remote_offer(
- webrtc::CreateSessionDescription(SdpType::kOffer, sdp));
+ CreateSessionDescription(SdpType::kOffer, sdp));
EXPECT_TRUE(DoSetRemoteDescription(std::move(remote_offer)));
EXPECT_EQ(PeerConnectionInterface::kHaveRemoteOffer, observer_.state_);
}
@@ -1013,7 +1003,7 @@
std::string sdp;
EXPECT_TRUE(answer->ToString(&sdp));
std::unique_ptr<SessionDescriptionInterface> new_answer(
- webrtc::CreateSessionDescription(SdpType::kAnswer, sdp));
+ CreateSessionDescription(SdpType::kAnswer, sdp));
EXPECT_TRUE(DoSetLocalDescription(std::move(new_answer)));
EXPECT_EQ(PeerConnectionInterface::kStable, observer_.state_);
}
@@ -1025,7 +1015,7 @@
std::string sdp;
EXPECT_TRUE(answer->ToString(&sdp));
std::unique_ptr<SessionDescriptionInterface> pr_answer(
- webrtc::CreateSessionDescription(SdpType::kPrAnswer, sdp));
+ CreateSessionDescription(SdpType::kPrAnswer, sdp));
EXPECT_TRUE(DoSetLocalDescription(std::move(pr_answer)));
EXPECT_EQ(PeerConnectionInterface::kHaveLocalPrAnswer, observer_.state_);
}
@@ -1050,7 +1040,7 @@
std::string sdp;
EXPECT_TRUE(offer->ToString(&sdp));
std::unique_ptr<SessionDescriptionInterface> new_offer(
- webrtc::CreateSessionDescription(SdpType::kOffer, sdp));
+ CreateSessionDescription(SdpType::kOffer, sdp));
EXPECT_TRUE(DoSetLocalDescription(std::move(new_offer)));
EXPECT_EQ(PeerConnectionInterface::kHaveLocalOffer, observer_.state_);
@@ -1060,7 +1050,7 @@
void CreateAnswerAsRemoteDescription(const std::string& sdp) {
std::unique_ptr<SessionDescriptionInterface> answer(
- webrtc::CreateSessionDescription(SdpType::kAnswer, sdp));
+ CreateSessionDescription(SdpType::kAnswer, sdp));
ASSERT_TRUE(answer);
EXPECT_TRUE(DoSetRemoteDescription(std::move(answer)));
EXPECT_EQ(PeerConnectionInterface::kStable, observer_.state_);
@@ -1068,12 +1058,12 @@
void CreatePrAnswerAndAnswerAsRemoteDescription(const std::string& sdp) {
std::unique_ptr<SessionDescriptionInterface> pr_answer(
- webrtc::CreateSessionDescription(SdpType::kPrAnswer, sdp));
+ CreateSessionDescription(SdpType::kPrAnswer, sdp));
ASSERT_TRUE(pr_answer);
EXPECT_TRUE(DoSetRemoteDescription(std::move(pr_answer)));
EXPECT_EQ(PeerConnectionInterface::kHaveRemotePrAnswer, observer_.state_);
std::unique_ptr<SessionDescriptionInterface> answer(
- webrtc::CreateSessionDescription(SdpType::kAnswer, sdp));
+ CreateSessionDescription(SdpType::kAnswer, sdp));
ASSERT_TRUE(answer);
EXPECT_TRUE(DoSetRemoteDescription(std::move(answer)));
EXPECT_EQ(PeerConnectionInterface::kStable, observer_.state_);
@@ -1117,8 +1107,8 @@
std::string mediastream_id = kStreams[0];
- rtc::scoped_refptr<webrtc::MediaStreamInterface> stream(
- webrtc::MediaStream::Create(mediastream_id));
+ rtc::scoped_refptr<MediaStreamInterface> stream(
+ MediaStream::Create(mediastream_id));
reference_collection_->AddStream(stream);
if (number_of_audio_tracks > 0) {
@@ -1142,22 +1132,20 @@
}
return std::unique_ptr<SessionDescriptionInterface>(
- webrtc::CreateSessionDescription(SdpType::kOffer, sdp_ms1));
+ CreateSessionDescription(SdpType::kOffer, sdp_ms1));
}
void AddAudioTrack(const std::string& track_id,
MediaStreamInterface* stream) {
- rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
- webrtc::AudioTrack::Create(track_id, nullptr));
+ rtc::scoped_refptr<AudioTrackInterface> audio_track(
+ AudioTrack::Create(track_id, nullptr));
ASSERT_TRUE(stream->AddTrack(audio_track));
}
void AddVideoTrack(const std::string& track_id,
MediaStreamInterface* stream) {
- rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track(
- webrtc::VideoTrack::Create(track_id,
- webrtc::FakeVideoTrackSource::Create(),
- rtc::Thread::Current()));
+ rtc::scoped_refptr<VideoTrackInterface> video_track(VideoTrack::Create(
+ track_id, FakeVideoTrackSource::Create(), rtc::Thread::Current()));
ASSERT_TRUE(stream->AddTrack(video_track));
}
@@ -1217,7 +1205,7 @@
std::string sdp;
EXPECT_TRUE((*desc)->ToString(&sdp));
std::unique_ptr<SessionDescriptionInterface> remote_offer(
- webrtc::CreateSessionDescription(SdpType::kOffer, sdp));
+ CreateSessionDescription(SdpType::kOffer, sdp));
EXPECT_TRUE(DoSetRemoteDescription(std::move(remote_offer)));
EXPECT_EQ(PeerConnectionInterface::kHaveRemoteOffer, observer_.state_);
}
@@ -1230,7 +1218,7 @@
std::string sdp;
EXPECT_TRUE((*desc)->ToString(&sdp));
std::unique_ptr<SessionDescriptionInterface> new_offer(
- webrtc::CreateSessionDescription(SdpType::kOffer, sdp));
+ CreateSessionDescription(SdpType::kOffer, sdp));
EXPECT_TRUE(DoSetLocalDescription(std::move(new_offer)));
EXPECT_EQ(PeerConnectionInterface::kHaveLocalOffer, observer_.state_);
@@ -1266,13 +1254,13 @@
rtc::SocketServer* socket_server() const { return vss_.get(); }
- webrtc::test::ScopedKeyValueConfig field_trials_;
+ test::ScopedKeyValueConfig field_trials_;
std::unique_ptr<rtc::VirtualSocketServer> vss_;
rtc::AutoSocketServerThread main_;
rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
cricket::FakePortAllocator* port_allocator_ = nullptr;
FakeRTCCertificateGenerator* fake_certificate_generator_ = nullptr;
- rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pc_factory_;
+ rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory_;
rtc::scoped_refptr<PeerConnectionInterface> pc_;
MockPeerConnectionObserver observer_;
rtc::scoped_refptr<StreamCollection> reference_collection_;
@@ -1392,22 +1380,19 @@
config.prune_turn_ports = true;
// Create the PC factory and PC with the above config.
- rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pc_factory(
- webrtc::CreatePeerConnectionFactory(
+ rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory(
+ CreatePeerConnectionFactory(
rtc::Thread::Current(), rtc::Thread::Current(),
rtc::Thread::Current(), fake_audio_capture_module_,
- webrtc::CreateBuiltinAudioEncoderFactory(),
- webrtc::CreateBuiltinAudioDecoderFactory(),
- std::make_unique<webrtc::VideoEncoderFactoryTemplate<
- webrtc::LibvpxVp8EncoderTemplateAdapter,
- webrtc::LibvpxVp9EncoderTemplateAdapter,
- webrtc::OpenH264EncoderTemplateAdapter,
- webrtc::LibaomAv1EncoderTemplateAdapter>>(),
- std::make_unique<webrtc::VideoDecoderFactoryTemplate<
- webrtc::LibvpxVp8DecoderTemplateAdapter,
- webrtc::LibvpxVp9DecoderTemplateAdapter,
- webrtc::OpenH264DecoderTemplateAdapter,
- webrtc::Dav1dDecoderTemplateAdapter>>(),
+ CreateBuiltinAudioEncoderFactory(),
+ CreateBuiltinAudioDecoderFactory(),
+ std::make_unique<VideoEncoderFactoryTemplate<
+ LibvpxVp8EncoderTemplateAdapter, LibvpxVp9EncoderTemplateAdapter,
+ OpenH264EncoderTemplateAdapter,
+ LibaomAv1EncoderTemplateAdapter>>(),
+ std::make_unique<VideoDecoderFactoryTemplate<
+ LibvpxVp8DecoderTemplateAdapter, LibvpxVp9DecoderTemplateAdapter,
+ OpenH264DecoderTemplateAdapter, Dav1dDecoderTemplateAdapter>>(),
nullptr /* audio_mixer */, nullptr /* audio_processing */));
PeerConnectionDependencies pc_dependencies(&observer_);
pc_dependencies.allocator = std::move(port_allocator);
@@ -1424,7 +1409,7 @@
EXPECT_TRUE(raw_port_allocator->flags() & cricket::PORTALLOCATOR_DISABLE_TCP);
EXPECT_TRUE(raw_port_allocator->flags() &
cricket::PORTALLOCATOR_DISABLE_COSTLY_NETWORKS);
- EXPECT_EQ(webrtc::PRUNE_BASED_ON_PRIORITY,
+ EXPECT_EQ(PRUNE_BASED_ON_PRIORITY,
raw_port_allocator->turn_port_prune_policy());
}
@@ -1446,8 +1431,7 @@
TEST_P(PeerConnectionInterfaceTest, GetConfigurationAfterSetConfiguration) {
PeerConnectionInterface::RTCConfiguration starting_config;
starting_config.sdp_semantics = sdp_semantics_;
- starting_config.bundle_policy =
- webrtc::PeerConnection::kBundlePolicyMaxBundle;
+ starting_config.bundle_policy = PeerConnection::kBundlePolicyMaxBundle;
CreatePeerConnection(starting_config);
PeerConnectionInterface::RTCConfiguration config = pc_->GetConfiguration();
@@ -1978,7 +1962,7 @@
RTCConfiguration rtc_config;
CreatePeerConnection(rtc_config);
- webrtc::DataChannelInit config;
+ DataChannelInit config;
auto channel = pc_->CreateDataChannelOrError("1", &config);
EXPECT_TRUE(channel.ok());
EXPECT_TRUE(channel.value()->reliable());
@@ -2010,7 +1994,7 @@
RTCConfiguration rtc_config;
CreatePeerConnection(rtc_config);
pc_->Close();
- webrtc::DataChannelInit config;
+ DataChannelInit config;
auto ret = pc_->CreateDataChannelOrError("1", &config);
ASSERT_FALSE(ret.ok());
EXPECT_EQ(ret.error().type(), RTCErrorType::INVALID_STATE);
@@ -2022,7 +2006,7 @@
RTCConfiguration rtc_config;
CreatePeerConnection(rtc_config);
- webrtc::DataChannelInit config;
+ DataChannelInit config;
config.maxRetransmitTime = -1;
config.maxRetransmits = -1;
auto channel = pc_->CreateDataChannelOrError("1", &config);
@@ -2037,7 +2021,7 @@
CreatePeerConnection(rtc_config);
std::string label = "test";
- webrtc::DataChannelInit config;
+ DataChannelInit config;
config.maxRetransmits = 0;
config.maxRetransmitTime = 0;
@@ -2052,7 +2036,7 @@
RTCConfiguration rtc_config;
CreatePeerConnection(rtc_config);
- webrtc::DataChannelInit config;
+ DataChannelInit config;
config.id = 1;
config.negotiated = true;
@@ -2106,7 +2090,7 @@
std::string sdp;
EXPECT_TRUE(pc_->local_description()->ToString(&sdp));
std::unique_ptr<SessionDescriptionInterface> answer(
- webrtc::CreateSessionDescription(SdpType::kAnswer, sdp));
+ CreateSessionDescription(SdpType::kAnswer, sdp));
ASSERT_TRUE(answer);
cricket::ContentInfo* data_info =
cricket::GetFirstDataContent(answer->description());
@@ -2125,8 +2109,7 @@
AddAudioTrack("audio_label");
AddVideoTrack("video_label");
std::unique_ptr<SessionDescriptionInterface> desc(
- webrtc::CreateSessionDescription(SdpType::kOffer,
- webrtc::kFireFoxSdpOffer, nullptr));
+ CreateSessionDescription(SdpType::kOffer, kFireFoxSdpOffer, nullptr));
EXPECT_TRUE(DoSetSessionDescription(std::move(desc), false));
CreateAnswerAsLocalDescription();
ASSERT_TRUE(pc_->local_description() != nullptr);
@@ -2163,8 +2146,7 @@
EXPECT_EQ_WAIT(1, fake_certificate_generator_->generated_certificates(),
kTimeout);
std::unique_ptr<SessionDescriptionInterface> desc(
- webrtc::CreateSessionDescription(SdpType::kOffer, kDtlsSdesFallbackSdp,
- nullptr));
+ CreateSessionDescription(SdpType::kOffer, kDtlsSdesFallbackSdp, nullptr));
EXPECT_FALSE(DoSetSessionDescription(std::move(desc), /*local=*/false));
}
@@ -2177,18 +2159,17 @@
CreateOfferAsLocalDescription();
const char* answer_sdp = (sdp_semantics_ == SdpSemantics::kPlanB_DEPRECATED
- ? webrtc::kAudioSdpPlanB
- : webrtc::kAudioSdpUnifiedPlan);
+ ? kAudioSdpPlanB
+ : kAudioSdpUnifiedPlan);
std::unique_ptr<SessionDescriptionInterface> answer(
- webrtc::CreateSessionDescription(SdpType::kAnswer, answer_sdp, nullptr));
+ CreateSessionDescription(SdpType::kAnswer, answer_sdp, nullptr));
EXPECT_TRUE(DoSetSessionDescription(std::move(answer), false));
- const char* reoffer_sdp =
- (sdp_semantics_ == SdpSemantics::kPlanB_DEPRECATED
- ? webrtc::kAudioSdpWithUnsupportedCodecsPlanB
- : webrtc::kAudioSdpWithUnsupportedCodecsUnifiedPlan);
+ const char* reoffer_sdp = (sdp_semantics_ == SdpSemantics::kPlanB_DEPRECATED
+ ? kAudioSdpWithUnsupportedCodecsPlanB
+ : kAudioSdpWithUnsupportedCodecsUnifiedPlan);
std::unique_ptr<SessionDescriptionInterface> updated_offer(
- webrtc::CreateSessionDescription(SdpType::kOffer, reoffer_sdp, nullptr));
+ CreateSessionDescription(SdpType::kOffer, reoffer_sdp, nullptr));
EXPECT_TRUE(DoSetSessionDescription(std::move(updated_offer), false));
CreateAnswerAsLocalDescription();
}
@@ -2275,12 +2256,11 @@
config.prune_turn_ports = false;
CreatePeerConnection(config);
config = pc_->GetConfiguration();
- EXPECT_EQ(webrtc::NO_PRUNE, port_allocator_->turn_port_prune_policy());
+ EXPECT_EQ(NO_PRUNE, port_allocator_->turn_port_prune_policy());
config.prune_turn_ports = true;
EXPECT_TRUE(pc_->SetConfiguration(config).ok());
- EXPECT_EQ(webrtc::PRUNE_BASED_ON_PRIORITY,
- port_allocator_->turn_port_prune_policy());
+ EXPECT_EQ(PRUNE_BASED_ON_PRIORITY, port_allocator_->turn_port_prune_policy());
}
// Test that the ice check interval can be changed. This does not verify that
@@ -2549,12 +2529,12 @@
std::string sdp;
ASSERT_TRUE(pc_->remote_description()->ToString(&sdp));
std::unique_ptr<SessionDescriptionInterface> remote_offer(
- webrtc::CreateSessionDescription(SdpType::kOffer, sdp));
+ CreateSessionDescription(SdpType::kOffer, sdp));
EXPECT_FALSE(DoSetRemoteDescription(std::move(remote_offer)));
ASSERT_TRUE(pc_->local_description()->ToString(&sdp));
std::unique_ptr<SessionDescriptionInterface> local_offer(
- webrtc::CreateSessionDescription(SdpType::kOffer, sdp));
+ CreateSessionDescription(SdpType::kOffer, sdp));
EXPECT_FALSE(DoSetLocalDescription(std::move(local_offer)));
}
@@ -2614,10 +2594,10 @@
reference_collection_.get()));
rtc::scoped_refptr<AudioTrackInterface> audio_track2 =
observer_.remote_streams()->at(0)->GetAudioTracks()[1];
- EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, audio_track2->state());
+ EXPECT_EQ(MediaStreamTrackInterface::kLive, audio_track2->state());
rtc::scoped_refptr<VideoTrackInterface> video_track2 =
observer_.remote_streams()->at(0)->GetVideoTracks()[1];
- EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, video_track2->state());
+ EXPECT_EQ(MediaStreamTrackInterface::kLive, video_track2->state());
// Remove the extra audio and video tracks.
std::unique_ptr<SessionDescriptionInterface> desc_ms2 =
@@ -2631,10 +2611,10 @@
EXPECT_TRUE(CompareStreamCollections(observer_.remote_streams(),
reference_collection_.get()));
// Track state may be updated asynchronously.
- EXPECT_EQ_WAIT(webrtc::MediaStreamTrackInterface::kEnded,
- audio_track2->state(), kTimeout);
- EXPECT_EQ_WAIT(webrtc::MediaStreamTrackInterface::kEnded,
- video_track2->state(), kTimeout);
+ EXPECT_EQ_WAIT(MediaStreamTrackInterface::kEnded, audio_track2->state(),
+ kTimeout);
+ EXPECT_EQ_WAIT(MediaStreamTrackInterface::kEnded, video_track2->state(),
+ kTimeout);
}
// This tests that remote tracks are ended if a local session description is set
@@ -2652,7 +2632,7 @@
rtc::scoped_refptr<MediaStreamTrackInterface> remote_audio =
audio_receiver->track();
- EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, remote_audio->state());
+ EXPECT_EQ(MediaStreamTrackInterface::kLive, remote_audio->state());
rtc::scoped_refptr<MediaStreamTrackInterface> remote_video =
video_receiver->track();
EXPECT_EQ(MediaStreamTrackInterface::kLive, remote_video->state());
@@ -2696,8 +2676,8 @@
remote_stream->RemoveTrack(remote_stream->GetAudioTracks()[0]);
std::unique_ptr<SessionDescriptionInterface> local_answer(
- webrtc::CreateSessionDescription(SdpType::kAnswer,
- GetSdpStringWithStream1(), nullptr));
+ CreateSessionDescription(SdpType::kAnswer, GetSdpStringWithStream1(),
+ nullptr));
cricket::ContentInfo* video_info =
local_answer->description()->GetContentByName("video");
video_info->rejected = true;
@@ -2986,9 +2966,9 @@
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
// Grab a copy of the offer before it gets passed into the PC.
std::unique_ptr<SessionDescriptionInterface> modified_offer =
- webrtc::CreateSessionDescription(
- webrtc::SdpType::kOffer, offer->session_id(),
- offer->session_version(), offer->description()->Clone());
+ CreateSessionDescription(SdpType::kOffer, offer->session_id(),
+ offer->session_version(),
+ offer->description()->Clone());
EXPECT_TRUE(DoSetLocalDescription(std::move(offer)));
auto senders = pc_->GetSenders();
@@ -3044,8 +3024,8 @@
EXPECT_TRUE(ContainsSender(senders, kVideoTracks[0], kStreams[0]));
// 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]));
+ rtc::scoped_refptr<MediaStreamInterface> stream_1(
+ 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.get());
@@ -3166,9 +3146,9 @@
EXPECT_TRUE(pc_->SetConfiguration(config).ok());
// Do ICE restart for the first m= section, initiated by remote peer.
- std::unique_ptr<webrtc::SessionDescriptionInterface> remote_offer(
- webrtc::CreateSessionDescription(SdpType::kOffer,
- GetSdpStringWithStream1(), nullptr));
+ std::unique_ptr<SessionDescriptionInterface> remote_offer(
+ CreateSessionDescription(SdpType::kOffer, GetSdpStringWithStream1(),
+ nullptr));
ASSERT_TRUE(remote_offer);
remote_offer->description()->transport_infos()[0].description.ice_ufrag =
"modified";
@@ -3214,7 +3194,7 @@
// Set remote pranswer.
std::unique_ptr<SessionDescriptionInterface> remote_pranswer(
- webrtc::CreateSessionDescription(SdpType::kPrAnswer, sdp));
+ CreateSessionDescription(SdpType::kPrAnswer, sdp));
SessionDescriptionInterface* remote_pranswer_ptr = remote_pranswer.get();
EXPECT_TRUE(DoSetRemoteDescription(std::move(remote_pranswer)));
EXPECT_EQ(local_offer_ptr, pc_->pending_local_description());
@@ -3224,7 +3204,7 @@
// Set remote answer.
std::unique_ptr<SessionDescriptionInterface> remote_answer(
- webrtc::CreateSessionDescription(SdpType::kAnswer, sdp));
+ CreateSessionDescription(SdpType::kAnswer, sdp));
SessionDescriptionInterface* remote_answer_ptr = remote_answer.get();
EXPECT_TRUE(DoSetRemoteDescription(std::move(remote_answer)));
EXPECT_EQ(nullptr, pc_->pending_local_description());
@@ -3234,7 +3214,7 @@
// Set remote offer.
std::unique_ptr<SessionDescriptionInterface> remote_offer(
- webrtc::CreateSessionDescription(SdpType::kOffer, sdp));
+ CreateSessionDescription(SdpType::kOffer, sdp));
SessionDescriptionInterface* remote_offer_ptr = remote_offer.get();
EXPECT_TRUE(DoSetRemoteDescription(std::move(remote_offer)));
EXPECT_EQ(remote_offer_ptr, pc_->pending_remote_description());
@@ -3244,7 +3224,7 @@
// Set local pranswer.
std::unique_ptr<SessionDescriptionInterface> local_pranswer(
- webrtc::CreateSessionDescription(SdpType::kPrAnswer, sdp));
+ CreateSessionDescription(SdpType::kPrAnswer, sdp));
SessionDescriptionInterface* local_pranswer_ptr = local_pranswer.get();
EXPECT_TRUE(DoSetLocalDescription(std::move(local_pranswer)));
EXPECT_EQ(remote_offer_ptr, pc_->pending_remote_description());
@@ -3254,7 +3234,7 @@
// Set local answer.
std::unique_ptr<SessionDescriptionInterface> local_answer(
- webrtc::CreateSessionDescription(SdpType::kAnswer, sdp));
+ CreateSessionDescription(SdpType::kAnswer, sdp));
SessionDescriptionInterface* local_answer_ptr = local_answer.get();
EXPECT_TRUE(DoSetLocalDescription(std::move(local_answer)));
EXPECT_EQ(nullptr, pc_->pending_remote_description());
@@ -3273,9 +3253,8 @@
// The RtcEventLog will be reset when the PeerConnection is closed.
pc_->Close();
- EXPECT_FALSE(
- pc_->StartRtcEventLog(std::make_unique<webrtc::RtcEventLogOutputNull>(),
- webrtc::RtcEventLog::kImmediateOutput));
+ EXPECT_FALSE(pc_->StartRtcEventLog(std::make_unique<RtcEventLogOutputNull>(),
+ RtcEventLog::kImmediateOutput));
pc_->StopRtcEventLog();
}
diff --git a/pc/peer_connection_media_unittest.cc b/pc/peer_connection_media_unittest.cc
index f08474e..796520f 100644
--- a/pc/peer_connection_media_unittest.cc
+++ b/pc/peer_connection_media_unittest.cc
@@ -82,9 +82,9 @@
using ::testing::Values;
cricket::MediaSendChannelInterface* SendChannelInternal(
- rtc::scoped_refptr<webrtc::RtpTransceiverInterface> transceiver) {
- auto transceiver_with_internal = static_cast<rtc::RefCountedObject<
- webrtc::RtpTransceiverProxyWithInternal<webrtc::RtpTransceiver>>*>(
+ rtc::scoped_refptr<RtpTransceiverInterface> transceiver) {
+ auto transceiver_with_internal = static_cast<
+ rtc::RefCountedObject<RtpTransceiverProxyWithInternal<RtpTransceiver>>*>(
transceiver.get());
auto transceiver_internal =
static_cast<RtpTransceiver*>(transceiver_with_internal->internal());
@@ -92,9 +92,9 @@
}
cricket::MediaReceiveChannelInterface* ReceiveChannelInternal(
- rtc::scoped_refptr<webrtc::RtpTransceiverInterface> transceiver) {
- auto transceiver_with_internal = static_cast<rtc::RefCountedObject<
- webrtc::RtpTransceiverProxyWithInternal<webrtc::RtpTransceiver>>*>(
+ rtc::scoped_refptr<RtpTransceiverInterface> transceiver) {
+ auto transceiver_with_internal = static_cast<
+ rtc::RefCountedObject<RtpTransceiverProxyWithInternal<RtpTransceiver>>*>(
transceiver.get());
auto transceiver_internal =
static_cast<RtpTransceiver*>(transceiver_with_internal->internal());
@@ -102,22 +102,22 @@
}
cricket::FakeVideoMediaSendChannel* VideoMediaSendChannel(
- rtc::scoped_refptr<webrtc::RtpTransceiverInterface> transceiver) {
+ rtc::scoped_refptr<RtpTransceiverInterface> transceiver) {
return static_cast<cricket::FakeVideoMediaSendChannel*>(
SendChannelInternal(transceiver));
}
cricket::FakeVideoMediaReceiveChannel* VideoMediaReceiveChannel(
- rtc::scoped_refptr<webrtc::RtpTransceiverInterface> transceiver) {
+ rtc::scoped_refptr<RtpTransceiverInterface> transceiver) {
return static_cast<cricket::FakeVideoMediaReceiveChannel*>(
ReceiveChannelInternal(transceiver));
}
cricket::FakeVoiceMediaSendChannel* VoiceMediaSendChannel(
- rtc::scoped_refptr<webrtc::RtpTransceiverInterface> transceiver) {
+ rtc::scoped_refptr<RtpTransceiverInterface> transceiver) {
return static_cast<cricket::FakeVoiceMediaSendChannel*>(
SendChannelInternal(transceiver));
}
cricket::FakeVoiceMediaReceiveChannel* VoiceMediaReceiveChannel(
- rtc::scoped_refptr<webrtc::RtpTransceiverInterface> transceiver) {
+ rtc::scoped_refptr<RtpTransceiverInterface> transceiver) {
return static_cast<cricket::FakeVoiceMediaReceiveChannel*>(
ReceiveChannelInternal(transceiver));
}
@@ -254,7 +254,7 @@
return sdp_semantics_ == SdpSemantics::kUnifiedPlan;
}
- webrtc::test::ScopedKeyValueConfig field_trials_;
+ test::ScopedKeyValueConfig field_trials_;
std::unique_ptr<rtc::VirtualSocketServer> vss_;
rtc::AutoSocketServerThread main_;
const SdpSemantics sdp_semantics_;
@@ -1495,10 +1495,10 @@
}
template <typename C>
-bool CompareCodecs(const std::vector<webrtc::RtpCodecCapability>& capabilities,
+bool CompareCodecs(const std::vector<RtpCodecCapability>& capabilities,
const std::vector<C>& codecs) {
bool capability_has_rtx =
- absl::c_any_of(capabilities, [](const webrtc::RtpCodecCapability& codec) {
+ absl::c_any_of(capabilities, [](const RtpCodecCapability& codec) {
return codec.name == cricket::kRtxCodecName;
});
bool codecs_has_rtx = absl::c_any_of(codecs, [](const C& codec) {
@@ -1510,16 +1510,16 @@
codecs, std::back_inserter(codecs_no_rtx),
[](const C& codec) { return codec.name != cricket::kRtxCodecName; });
- std::vector<webrtc::RtpCodecCapability> capabilities_no_rtx;
+ std::vector<RtpCodecCapability> capabilities_no_rtx;
absl::c_copy_if(capabilities, std::back_inserter(capabilities_no_rtx),
- [](const webrtc::RtpCodecCapability& codec) {
+ [](const RtpCodecCapability& codec) {
return codec.name != cricket::kRtxCodecName;
});
return capability_has_rtx == codecs_has_rtx &&
absl::c_equal(
capabilities_no_rtx, codecs_no_rtx,
- [](const webrtc::RtpCodecCapability& capability, const C& codec) {
+ [](const RtpCodecCapability& capability, const C& codec) {
return codec.MatchesRtpCodec(capability);
});
}
@@ -1538,9 +1538,9 @@
auto capabilities = caller->pc_factory()->GetRtpSenderCapabilities(
cricket::MediaType::MEDIA_TYPE_AUDIO);
- std::vector<webrtc::RtpCodecCapability> codecs;
+ std::vector<RtpCodecCapability> codecs;
absl::c_copy_if(capabilities.codecs, std::back_inserter(codecs),
- [](const webrtc::RtpCodecCapability& codec) {
+ [](const RtpCodecCapability& codec) {
return codec.name.find("_only_") != std::string::npos;
});
@@ -1561,9 +1561,9 @@
auto capabilities = caller->pc_factory()->GetRtpReceiverCapabilities(
cricket::MediaType::MEDIA_TYPE_AUDIO);
- std::vector<webrtc::RtpCodecCapability> codecs;
+ std::vector<RtpCodecCapability> codecs;
absl::c_copy_if(capabilities.codecs, std::back_inserter(codecs),
- [](const webrtc::RtpCodecCapability& codec) {
+ [](const RtpCodecCapability& codec) {
return codec.name.find("_only_") != std::string::npos;
});
@@ -1611,7 +1611,7 @@
auto codecs_only_rtx_red_fec = codecs;
auto it = std::remove_if(codecs_only_rtx_red_fec.begin(),
codecs_only_rtx_red_fec.end(),
- [](const webrtc::RtpCodecCapability& codec) {
+ [](const RtpCodecCapability& codec) {
return !(codec.name == cricket::kRtxCodecName ||
codec.name == cricket::kRedCodecName ||
codec.name == cricket::kUlpfecCodecName);
@@ -1651,7 +1651,7 @@
caller->pc_factory()
->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_AUDIO)
.codecs;
- std::vector<webrtc::RtpCodecCapability> empty_codecs = {};
+ std::vector<RtpCodecCapability> empty_codecs = {};
auto audio_transceiver = caller->pc()->GetTransceivers().front();
@@ -1706,7 +1706,7 @@
auto codecs_only_rtx_red_fec = codecs;
auto it = std::remove_if(codecs_only_rtx_red_fec.begin(),
codecs_only_rtx_red_fec.end(),
- [](const webrtc::RtpCodecCapability& codec) {
+ [](const RtpCodecCapability& codec) {
return !(codec.name == cricket::kRtxCodecName ||
codec.name == cricket::kRedCodecName ||
codec.name == cricket::kUlpfecCodecName);
@@ -1747,7 +1747,7 @@
->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO)
.codecs;
- std::vector<webrtc::RtpCodecCapability> empty_codecs = {};
+ std::vector<RtpCodecCapability> empty_codecs = {};
auto video_transceiver = caller->pc()->GetTransceivers().front();
@@ -1817,7 +1817,7 @@
auto video_codecs_vpx_rtx = sender_video_codecs;
auto it =
std::remove_if(video_codecs_vpx_rtx.begin(), video_codecs_vpx_rtx.end(),
- [](const webrtc::RtpCodecCapability& codec) {
+ [](const RtpCodecCapability& codec) {
return codec.name != cricket::kRtxCodecName &&
codec.name != cricket::kVp8CodecName &&
codec.name != cricket::kVp9CodecName;
@@ -1866,7 +1866,7 @@
auto video_codecs_vpx = video_codecs;
auto it = std::remove_if(video_codecs_vpx.begin(), video_codecs_vpx.end(),
- [](const webrtc::RtpCodecCapability& codec) {
+ [](const RtpCodecCapability& codec) {
return codec.name != cricket::kVp8CodecName &&
codec.name != cricket::kVp9CodecName;
});
@@ -1889,7 +1889,7 @@
auto recv_transceiver = callee->pc()->GetTransceivers().front();
auto video_codecs_vp8_rtx = video_codecs;
it = std::remove_if(video_codecs_vp8_rtx.begin(), video_codecs_vp8_rtx.end(),
- [](const webrtc::RtpCodecCapability& codec) {
+ [](const RtpCodecCapability& codec) {
bool r = codec.name != cricket::kVp8CodecName &&
codec.name != cricket::kRtxCodecName;
return r;
@@ -1936,7 +1936,7 @@
auto video_codecs_vpx = video_codecs;
auto it = std::remove_if(video_codecs_vpx.begin(), video_codecs_vpx.end(),
- [](const webrtc::RtpCodecCapability& codec) {
+ [](const RtpCodecCapability& codec) {
return codec.name != cricket::kVp8CodecName &&
codec.name != cricket::kVp9CodecName;
});
diff --git a/pc/peer_connection_rampup_tests.cc b/pc/peer_connection_rampup_tests.cc
index 545a1d5..0fd3c27 100644
--- a/pc/peer_connection_rampup_tests.cc
+++ b/pc/peer_connection_rampup_tests.cc
@@ -201,7 +201,7 @@
fake_network_managers_.emplace_back(fake_network_manager);
auto observer = std::make_unique<MockPeerConnectionObserver>();
- webrtc::PeerConnectionDependencies dependencies(observer.get());
+ PeerConnectionDependencies dependencies(observer.get());
cricket::BasicPortAllocator* port_allocator =
new cricket::BasicPortAllocator(fake_network_manager,
firewall_socket_factory_.get());
diff --git a/pc/peer_connection_rtp_unittest.cc b/pc/peer_connection_rtp_unittest.cc
index b93e592..1a97a4b 100644
--- a/pc/peer_connection_rtp_unittest.cc
+++ b/pc/peer_connection_rtp_unittest.cc
@@ -75,13 +75,13 @@
using ::testing::Values;
template <typename MethodFunctor>
-class OnSuccessObserver : public webrtc::SetRemoteDescriptionObserverInterface {
+class OnSuccessObserver : public SetRemoteDescriptionObserverInterface {
public:
explicit OnSuccessObserver(MethodFunctor on_success)
: on_success_(std::move(on_success)) {}
- // webrtc::SetRemoteDescriptionObserverInterface implementation.
- void OnSetRemoteDescriptionComplete(webrtc::RTCError error) override {
+ // SetRemoteDescriptionObserverInterface implementation.
+ void OnSetRemoteDescriptionComplete(RTCError error) override {
RTC_CHECK(error.ok());
on_success_();
}
@@ -113,7 +113,7 @@
Dav1dDecoderTemplateAdapter>>(),
nullptr /* audio_mixer */,
nullptr /* audio_processing */)) {
- webrtc::metrics::Reset();
+ metrics::Reset();
}
std::unique_ptr<PeerConnectionWrapper> CreatePeerConnection() {
@@ -201,7 +201,7 @@
}
};
-// These tests cover `webrtc::PeerConnectionObserver` callbacks firing upon
+// These tests cover `PeerConnectionObserver` callbacks firing upon
// setting the remote description.
TEST_P(PeerConnectionRtpTest, AddTrackWithoutStreamFiresOnAddTrack) {
@@ -934,8 +934,8 @@
auto caller = CreatePeerConnection();
auto callee = CreatePeerConnection();
- rtc::scoped_refptr<webrtc::MockSetSessionDescriptionObserver> observer =
- rtc::make_ref_counted<webrtc::MockSetSessionDescriptionObserver>();
+ rtc::scoped_refptr<MockSetSessionDescriptionObserver> observer =
+ rtc::make_ref_counted<MockSetSessionDescriptionObserver>();
auto offer = caller->CreateOfferAndSetAsLocal();
callee->pc()->SetRemoteDescription(observer.get(), offer.release());
diff --git a/pc/peer_connection_signaling_unittest.cc b/pc/peer_connection_signaling_unittest.cc
index 8ca59fc..aeba7ef 100644
--- a/pc/peer_connection_signaling_unittest.cc
+++ b/pc/peer_connection_signaling_unittest.cc
@@ -896,8 +896,8 @@
"m=bogus 9 FOO 0 8\r\n"
"c=IN IP4 0.0.0.0\r\n"
"a=mid:bogusmid\r\n";
- std::unique_ptr<webrtc::SessionDescriptionInterface> remote_description =
- webrtc::CreateSessionDescription(SdpType::kOffer, sdp, nullptr);
+ std::unique_ptr<SessionDescriptionInterface> remote_description =
+ CreateSessionDescription(SdpType::kOffer, sdp, nullptr);
EXPECT_TRUE(caller->SetRemoteDescription(std::move(remote_description)));
@@ -977,8 +977,8 @@
"a=ssrc-group:FEC-FR 1224551896 1953032773\r\n"
"a=ssrc:1224551896 cname:/exJcmhSLpyu9FgV\r\n"
"a=ssrc:1953032773 cname:/exJcmhSLpyu9FgV\r\n";
- std::unique_ptr<webrtc::SessionDescriptionInterface> remote_description =
- webrtc::CreateSessionDescription(SdpType::kOffer, sdp, nullptr);
+ std::unique_ptr<SessionDescriptionInterface> remote_description =
+ CreateSessionDescription(SdpType::kOffer, sdp, nullptr);
EXPECT_TRUE(caller->SetRemoteDescription(std::move(remote_description)));
@@ -1033,8 +1033,8 @@
"a=ssrc-group:FEC-FR 1224551896 1953032773\r\n"
"a=ssrc:1224551896 cname:/exJcmhSLpyu9FgV\r\n"
"a=ssrc:1953032773 cname:/exJcmhSLpyu9FgV\r\n";
- std::unique_ptr<webrtc::SessionDescriptionInterface> remote_description =
- webrtc::CreateSessionDescription(SdpType::kOffer, sdp, nullptr);
+ std::unique_ptr<SessionDescriptionInterface> remote_description =
+ CreateSessionDescription(SdpType::kOffer, sdp, nullptr);
EXPECT_TRUE(caller->SetRemoteDescription(std::move(remote_description)));
@@ -1104,8 +1104,8 @@
"a=rtcp-fb:102 nack\r\n"
"a=rtcp-fb:102 nack pli\r\n"
"a=ssrc:1224551896 cname:/exJcmhSLpyu9FgV\r\n";
- std::unique_ptr<webrtc::SessionDescriptionInterface> remote_description =
- webrtc::CreateSessionDescription(SdpType::kOffer, sdp, nullptr);
+ std::unique_ptr<SessionDescriptionInterface> remote_description =
+ CreateSessionDescription(SdpType::kOffer, sdp, nullptr);
EXPECT_FALSE(caller->SetRemoteDescription(std::move(remote_description)));
}
@@ -1339,8 +1339,8 @@
"a=rtcp-fb:102 nack\r\n"
"a=rtcp-fb:102 nack pli\r\n"
"a=ssrc:1224551896 cname:/exJcmhSLpyu9FgV\r\n";
- std::unique_ptr<webrtc::SessionDescriptionInterface> remote_description =
- webrtc::CreateSessionDescription(SdpType::kOffer, sdp, nullptr);
+ std::unique_ptr<SessionDescriptionInterface> remote_description =
+ CreateSessionDescription(SdpType::kOffer, sdp, nullptr);
EXPECT_TRUE(callee->SetRemoteDescription(std::move(remote_description)));
diff --git a/pc/peer_connection_simulcast_unittest.cc b/pc/peer_connection_simulcast_unittest.cc
index 6b6a96c..bffb5d9 100644
--- a/pc/peer_connection_simulcast_unittest.cc
+++ b/pc/peer_connection_simulcast_unittest.cc
@@ -220,7 +220,7 @@
: public PeerConnectionSimulcastTests,
public ::testing::WithParamInterface<int> {
protected:
- PeerConnectionSimulcastMetricsTests() { webrtc::metrics::Reset(); }
+ PeerConnectionSimulcastMetricsTests() { metrics::Reset(); }
};
#endif
diff --git a/pc/peer_connection_svc_integrationtest.cc b/pc/peer_connection_svc_integrationtest.cc
index 672f3ee..32ca451 100644
--- a/pc/peer_connection_svc_integrationtest.cc
+++ b/pc/peer_connection_svc_integrationtest.cc
@@ -37,14 +37,13 @@
: PeerConnectionIntegrationBaseTest(SdpSemantics::kUnifiedPlan) {}
RTCError SetCodecPreferences(
- rtc::scoped_refptr<webrtc::RtpTransceiverInterface> transceiver,
+ rtc::scoped_refptr<RtpTransceiverInterface> transceiver,
absl::string_view codec_name) {
- webrtc::RtpCapabilities capabilities =
+ RtpCapabilities capabilities =
caller()->pc_factory()->GetRtpSenderCapabilities(
cricket::MEDIA_TYPE_VIDEO);
std::vector<RtpCodecCapability> codecs;
- for (const webrtc::RtpCodecCapability& codec_capability :
- capabilities.codecs) {
+ for (const RtpCodecCapability& codec_capability : capabilities.codecs) {
if (codec_capability.name == codec_name)
codecs.push_back(codec_capability);
}
@@ -55,8 +54,8 @@
TEST_F(PeerConnectionSVCIntegrationTest, AddTransceiverAcceptsL1T1) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
- webrtc::RtpTransceiverInit init;
- webrtc::RtpEncodingParameters encoding_parameters;
+ RtpTransceiverInit init;
+ RtpEncodingParameters encoding_parameters;
encoding_parameters.scalability_mode = "L1T1";
init.send_encodings.push_back(encoding_parameters);
auto transceiver_or_error =
@@ -67,8 +66,8 @@
TEST_F(PeerConnectionSVCIntegrationTest, AddTransceiverAcceptsL3T3) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
- webrtc::RtpTransceiverInit init;
- webrtc::RtpEncodingParameters encoding_parameters;
+ RtpTransceiverInit init;
+ RtpEncodingParameters encoding_parameters;
encoding_parameters.scalability_mode = "L3T3";
init.send_encodings.push_back(encoding_parameters);
auto transceiver_or_error =
@@ -80,33 +79,32 @@
AddTransceiverRejectsUnknownScalabilityMode) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
- webrtc::RtpTransceiverInit init;
- webrtc::RtpEncodingParameters encoding_parameters;
+ RtpTransceiverInit init;
+ RtpEncodingParameters encoding_parameters;
encoding_parameters.scalability_mode = "FOOBAR";
init.send_encodings.push_back(encoding_parameters);
auto transceiver_or_error =
caller()->pc()->AddTransceiver(caller()->CreateLocalVideoTrack(), init);
EXPECT_FALSE(transceiver_or_error.ok());
EXPECT_EQ(transceiver_or_error.error().type(),
- webrtc::RTCErrorType::UNSUPPORTED_OPERATION);
+ RTCErrorType::UNSUPPORTED_OPERATION);
}
TEST_F(PeerConnectionSVCIntegrationTest, SetParametersAcceptsL1T3WithVP8) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
- webrtc::RtpCapabilities capabilities =
+ RtpCapabilities capabilities =
caller()->pc_factory()->GetRtpSenderCapabilities(
cricket::MEDIA_TYPE_VIDEO);
std::vector<RtpCodecCapability> vp8_codec;
- for (const webrtc::RtpCodecCapability& codec_capability :
- capabilities.codecs) {
+ for (const RtpCodecCapability& codec_capability : capabilities.codecs) {
if (codec_capability.name == cricket::kVp8CodecName)
vp8_codec.push_back(codec_capability);
}
- webrtc::RtpTransceiverInit init;
- webrtc::RtpEncodingParameters encoding_parameters;
+ RtpTransceiverInit init;
+ RtpEncodingParameters encoding_parameters;
init.send_encodings.push_back(encoding_parameters);
auto transceiver_or_error =
caller()->pc()->AddTransceiver(caller()->CreateLocalVideoTrack(), init);
@@ -114,7 +112,7 @@
auto transceiver = transceiver_or_error.MoveValue();
EXPECT_TRUE(transceiver->SetCodecPreferences(vp8_codec).ok());
- webrtc::RtpParameters parameters = transceiver->sender()->GetParameters();
+ RtpParameters parameters = transceiver->sender()->GetParameters();
ASSERT_EQ(parameters.encodings.size(), 1u);
parameters.encodings[0].scalability_mode = "L1T3";
auto result = transceiver->sender()->SetParameters(parameters);
@@ -125,8 +123,8 @@
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
- webrtc::RtpTransceiverInit init;
- webrtc::RtpEncodingParameters encoding_parameters;
+ RtpTransceiverInit init;
+ RtpEncodingParameters encoding_parameters;
init.send_encodings.push_back(encoding_parameters);
auto transceiver_or_error =
caller()->pc()->AddTransceiver(caller()->CreateLocalVideoTrack(), init);
@@ -134,12 +132,12 @@
auto transceiver = transceiver_or_error.MoveValue();
EXPECT_TRUE(SetCodecPreferences(transceiver, cricket::kVp8CodecName).ok());
- webrtc::RtpParameters parameters = transceiver->sender()->GetParameters();
+ RtpParameters parameters = transceiver->sender()->GetParameters();
ASSERT_EQ(parameters.encodings.size(), 1u);
parameters.encodings[0].scalability_mode = "L3T3";
auto result = transceiver->sender()->SetParameters(parameters);
EXPECT_FALSE(result.ok());
- EXPECT_EQ(result.type(), webrtc::RTCErrorType::INVALID_MODIFICATION);
+ EXPECT_EQ(result.type(), RTCErrorType::INVALID_MODIFICATION);
}
TEST_F(PeerConnectionSVCIntegrationTest,
@@ -147,8 +145,8 @@
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
- webrtc::RtpTransceiverInit init;
- webrtc::RtpEncodingParameters encoding_parameters;
+ RtpTransceiverInit init;
+ RtpEncodingParameters encoding_parameters;
init.send_encodings.push_back(encoding_parameters);
auto transceiver_or_error =
caller()->pc()->AddTransceiver(caller()->CreateLocalVideoTrack(), init);
@@ -159,7 +157,7 @@
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
- webrtc::RtpParameters parameters = transceiver->sender()->GetParameters();
+ RtpParameters parameters = transceiver->sender()->GetParameters();
ASSERT_EQ(parameters.encodings.size(), 1u);
parameters.encodings[0].scalability_mode = "L1T3";
auto result = transceiver->sender()->SetParameters(parameters);
@@ -171,8 +169,8 @@
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
- webrtc::RtpTransceiverInit init;
- webrtc::RtpEncodingParameters encoding_parameters;
+ RtpTransceiverInit init;
+ RtpEncodingParameters encoding_parameters;
init.send_encodings.push_back(encoding_parameters);
auto transceiver_or_error =
caller()->pc()->AddTransceiver(caller()->CreateLocalVideoTrack(), init);
@@ -183,7 +181,7 @@
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
- webrtc::RtpParameters parameters = transceiver->sender()->GetParameters();
+ RtpParameters parameters = transceiver->sender()->GetParameters();
ASSERT_EQ(parameters.encodings.size(), 1u);
parameters.encodings[0].scalability_mode = "L3T3";
auto result = transceiver->sender()->SetParameters(parameters);
@@ -195,8 +193,8 @@
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
- webrtc::RtpTransceiverInit init;
- webrtc::RtpEncodingParameters encoding_parameters;
+ RtpTransceiverInit init;
+ RtpEncodingParameters encoding_parameters;
init.send_encodings.push_back(encoding_parameters);
auto transceiver_or_error =
caller()->pc()->AddTransceiver(caller()->CreateLocalVideoTrack(), init);
@@ -207,12 +205,12 @@
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
- webrtc::RtpParameters parameters = transceiver->sender()->GetParameters();
+ RtpParameters parameters = transceiver->sender()->GetParameters();
ASSERT_EQ(parameters.encodings.size(), 1u);
parameters.encodings[0].scalability_mode = "L3T3";
auto result = transceiver->sender()->SetParameters(parameters);
EXPECT_FALSE(result.ok());
- EXPECT_EQ(result.type(), webrtc::RTCErrorType::INVALID_MODIFICATION);
+ EXPECT_EQ(result.type(), RTCErrorType::INVALID_MODIFICATION);
}
TEST_F(PeerConnectionSVCIntegrationTest,
@@ -220,8 +218,8 @@
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
- webrtc::RtpTransceiverInit init;
- webrtc::RtpEncodingParameters encoding_parameters;
+ RtpTransceiverInit init;
+ RtpEncodingParameters encoding_parameters;
init.send_encodings.push_back(encoding_parameters);
auto transceiver_or_error =
caller()->pc()->AddTransceiver(caller()->CreateLocalVideoTrack(), init);
@@ -232,27 +230,27 @@
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
- webrtc::RtpParameters parameters = transceiver->sender()->GetParameters();
+ RtpParameters parameters = transceiver->sender()->GetParameters();
ASSERT_EQ(parameters.encodings.size(), 1u);
parameters.encodings[0].scalability_mode = "FOOBAR";
auto result = transceiver->sender()->SetParameters(parameters);
EXPECT_FALSE(result.ok());
- EXPECT_EQ(result.type(), webrtc::RTCErrorType::INVALID_MODIFICATION);
+ EXPECT_EQ(result.type(), RTCErrorType::INVALID_MODIFICATION);
}
TEST_F(PeerConnectionSVCIntegrationTest, FallbackToL1Tx) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
- webrtc::RtpTransceiverInit init;
- webrtc::RtpEncodingParameters encoding_parameters;
+ RtpTransceiverInit init;
+ RtpEncodingParameters encoding_parameters;
init.send_encodings.push_back(encoding_parameters);
auto transceiver_or_error =
caller()->pc()->AddTransceiver(caller()->CreateLocalVideoTrack(), init);
ASSERT_TRUE(transceiver_or_error.ok());
auto caller_transceiver = transceiver_or_error.MoveValue();
- webrtc::RtpCapabilities capabilities =
+ RtpCapabilities capabilities =
caller()->pc_factory()->GetRtpSenderCapabilities(
cricket::MEDIA_TYPE_VIDEO);
std::vector<RtpCodecCapability> send_codecs = capabilities.codecs;
@@ -267,8 +265,7 @@
caller_transceiver->SetCodecPreferences(send_codecs);
// L3T3 should be supported by VP9
- webrtc::RtpParameters parameters =
- caller_transceiver->sender()->GetParameters();
+ RtpParameters parameters = caller_transceiver->sender()->GetParameters();
ASSERT_EQ(parameters.encodings.size(), 1u);
parameters.encodings[0].scalability_mode = "L3T3";
auto result = caller_transceiver->sender()->SetParameters(parameters);
diff --git a/pc/peer_connection_wrapper.cc b/pc/peer_connection_wrapper.cc
index 44f4256..557d0c8 100644
--- a/pc/peer_connection_wrapper.cc
+++ b/pc/peer_connection_wrapper.cc
@@ -339,8 +339,7 @@
return observer()->ice_connected_;
}
-rtc::scoped_refptr<const webrtc::RTCStatsReport>
-PeerConnectionWrapper::GetStats() {
+rtc::scoped_refptr<const RTCStatsReport> PeerConnectionWrapper::GetStats() {
auto callback = rtc::make_ref_counted<MockRTCStatsCollectorCallback>();
pc()->GetStats(callback.get());
EXPECT_TRUE_WAIT(callback->called(), kDefaultTimeout);
diff --git a/pc/rtc_stats_collector.cc b/pc/rtc_stats_collector.cc
index 0797ba2..2bac176 100644
--- a/pc/rtc_stats_collector.cc
+++ b/pc/rtc_stats_collector.cc
@@ -336,7 +336,7 @@
std::map<std::string, double>
QualityLimitationDurationToRTCQualityLimitationDuration(
- std::map<webrtc::QualityLimitationReason, int64_t> durations_ms) {
+ std::map<QualityLimitationReason, int64_t> durations_ms) {
std::map<std::string, double> result;
// The internal duration is defined in milliseconds while the spec defines
// the value in seconds:
@@ -513,7 +513,7 @@
std::unique_ptr<RTCAudioPlayoutStats> CreateAudioPlayoutStats(
const AudioDeviceModule::Stats& audio_device_stats,
- webrtc::Timestamp timestamp) {
+ Timestamp timestamp) {
auto stats = std::make_unique<RTCAudioPlayoutStats>(
/*id=*/kAudioPlayoutSingletonId, timestamp);
stats->synthesized_samples_duration =
diff --git a/pc/rtc_stats_collector.h b/pc/rtc_stats_collector.h
index e94d239..4c68e77 100644
--- a/pc/rtc_stats_collector.h
+++ b/pc/rtc_stats_collector.h
@@ -317,7 +317,7 @@
uint32_t data_channels_closed;
// Identifies channels that have been opened, whose internal id is stored in
// the set until they have been fully closed.
- webrtc::flat_set<int> opened_data_channels;
+ flat_set<int> opened_data_channels;
};
InternalRecord internal_record_;
};
diff --git a/pc/rtc_stats_collector_unittest.cc b/pc/rtc_stats_collector_unittest.cc
index 37821ac..055be6f 100644
--- a/pc/rtc_stats_collector_unittest.cc
+++ b/pc/rtc_stats_collector_unittest.cc
@@ -263,9 +263,9 @@
std::string kind() const override {
return MediaStreamTrackInterface::kAudioKind;
}
- webrtc::AudioSourceInterface* GetSource() const override { return nullptr; }
- void AddSink(webrtc::AudioTrackSinkInterface* sink) override {}
- void RemoveSink(webrtc::AudioTrackSinkInterface* sink) override {}
+ AudioSourceInterface* GetSource() const override { return nullptr; }
+ void AddSink(AudioTrackSinkInterface* sink) override {}
+ void RemoveSink(AudioTrackSinkInterface* sink) override {}
bool GetSignalLevel(int* level) override { return false; }
rtc::scoped_refptr<AudioProcessorInterface> GetAudioProcessor() override {
return processor_;
@@ -2030,7 +2030,7 @@
EXPECT_TRUE(report->Get(*expected_pair.transport_id));
// Set bandwidth and "GetStats" again.
- webrtc::Call::Stats call_stats;
+ Call::Stats call_stats;
const int kSendBandwidth = 888;
call_stats.send_bandwidth_bps = kSendBandwidth;
const int kRecvBandwidth = 999;
@@ -2339,12 +2339,9 @@
video_media_info.receivers[0].key_frames_decoded = 3;
video_media_info.receivers[0].frames_dropped = 13;
video_media_info.receivers[0].qp_sum = absl::nullopt;
- video_media_info.receivers[0].total_decode_time =
- webrtc::TimeDelta::Seconds(9);
- video_media_info.receivers[0].total_processing_delay =
- webrtc::TimeDelta::Millis(600);
- video_media_info.receivers[0].total_assembly_time =
- webrtc::TimeDelta::Millis(500);
+ video_media_info.receivers[0].total_decode_time = TimeDelta::Seconds(9);
+ video_media_info.receivers[0].total_processing_delay = TimeDelta::Millis(600);
+ video_media_info.receivers[0].total_assembly_time = TimeDelta::Millis(500);
video_media_info.receivers[0].frames_assembled_from_multiple_packets = 23;
video_media_info.receivers[0].total_inter_frame_delay = 0.123;
video_media_info.receivers[0].total_squared_inter_frame_delay = 0.00456;
@@ -2617,12 +2614,12 @@
video_media_info.senders[0].key_frames_encoded = 3;
video_media_info.senders[0].total_encode_time_ms = 9000;
video_media_info.senders[0].total_encoded_bytes_target = 1234;
- video_media_info.senders[0].total_packet_send_delay =
- webrtc::TimeDelta::Seconds(10);
+ video_media_info.senders[0].total_packet_send_delay = TimeDelta::Seconds(10);
video_media_info.senders[0].quality_limitation_reason =
QualityLimitationReason::kBandwidth;
- video_media_info.senders[0].quality_limitation_durations_ms
- [webrtc::QualityLimitationReason::kBandwidth] = 300;
+ video_media_info.senders[0]
+ .quality_limitation_durations_ms[QualityLimitationReason::kBandwidth] =
+ 300;
video_media_info.senders[0].quality_limitation_resolution_changes = 56u;
video_media_info.senders[0].qp_sum = absl::nullopt;
video_media_info.senders[0].content_type = VideoContentType::UNSPECIFIED;
diff --git a/pc/rtc_stats_traversal_unittest.cc b/pc/rtc_stats_traversal_unittest.cc
index 72ad255..8205ebe 100644
--- a/pc/rtc_stats_traversal_unittest.cc
+++ b/pc/rtc_stats_traversal_unittest.cc
@@ -47,7 +47,7 @@
for (const RTCStats* start_node : start_nodes) {
start_ids.push_back(start_node->id());
}
- result_ = webrtc::TakeReferencedStats(initial_report_, start_ids);
+ result_ = ::webrtc::TakeReferencedStats(initial_report_, start_ids);
}
void EXPECT_VISITED(const RTCStats* stats) {
diff --git a/pc/rtp_sender.cc b/pc/rtp_sender.cc
index cdae159..b0c32ef 100644
--- a/pc/rtp_sender.cc
+++ b/pc/rtp_sender.cc
@@ -115,13 +115,13 @@
if (!signaling_thread_->IsCurrent()) {
signaling_thread_->PostTask(
[callback = std::move(callback_), error]() mutable {
- webrtc::InvokeSetParametersCallback(callback, error);
+ InvokeSetParametersCallback(callback, error);
});
callback_ = nullptr;
return;
}
- webrtc::InvokeSetParametersCallback(callback_, error);
+ InvokeSetParametersCallback(callback_, error);
callback_ = nullptr;
}
@@ -243,7 +243,7 @@
"Attempted to set an unimplemented parameter of RtpParameters.");
RTC_LOG(LS_ERROR) << error.message() << " ("
<< ::webrtc::ToString(error.type()) << ")";
- webrtc::InvokeSetParametersCallback(callback, error);
+ InvokeSetParametersCallback(callback, error);
return;
}
if (!media_channel_ || !ssrc_) {
@@ -252,7 +252,7 @@
if (result.ok()) {
init_parameters_ = parameters;
}
- webrtc::InvokeSetParametersCallback(callback, result);
+ InvokeSetParametersCallback(callback, result);
return;
}
auto task = [&, callback = std::move(callback),
@@ -268,13 +268,13 @@
RTCError result = cricket::CheckRtpParametersInvalidModificationAndValues(
old_parameters, rtp_parameters);
if (!result.ok()) {
- webrtc::InvokeSetParametersCallback(callback, result);
+ InvokeSetParametersCallback(callback, result);
return;
}
result = CheckCodecParameters(rtp_parameters);
if (!result.ok()) {
- webrtc::InvokeSetParametersCallback(callback, result);
+ InvokeSetParametersCallback(callback, result);
return;
}
@@ -389,7 +389,7 @@
TRACE_EVENT0("webrtc", "RtpSenderBase::SetParametersAsync");
RTCError result = CheckSetParameters(parameters);
if (!result.ok()) {
- webrtc::InvokeSetParametersCallback(callback, result);
+ InvokeSetParametersCallback(callback, result);
return;
}
@@ -399,7 +399,7 @@
signaling_thread_,
[this, callback = std::move(callback)](RTCError error) mutable {
last_transaction_id_.reset();
- webrtc::InvokeSetParametersCallback(callback, error);
+ InvokeSetParametersCallback(callback, error);
}),
false);
}
diff --git a/pc/rtp_sender.h b/pc/rtp_sender.h
index d29c376..26adceb 100644
--- a/pc/rtp_sender.h
+++ b/pc/rtp_sender.h
@@ -87,7 +87,7 @@
// Additional checks that are specific to the current codec settings
virtual RTCError CheckCodecParameters(const RtpParameters& parameters) {
- return webrtc::RTCError::OK();
+ return RTCError::OK();
}
// Returns an ID that changes every time SetTrack() is called, but
diff --git a/pc/rtp_sender_receiver_unittest.cc b/pc/rtp_sender_receiver_unittest.cc
index 3092e53..4387aed 100644
--- a/pc/rtp_sender_receiver_unittest.cc
+++ b/pc/rtp_sender_receiver_unittest.cc
@@ -105,7 +105,7 @@
: network_thread_(rtc::Thread::Current()),
worker_thread_(rtc::Thread::Current()),
video_bitrate_allocator_factory_(
- webrtc::CreateBuiltinVideoBitrateAllocatorFactory()),
+ CreateBuiltinVideoBitrateAllocatorFactory()),
// Create fake media engine/etc. so we can create channels to use to
// test RtpSenders/RtpReceivers.
media_engine_(std::make_unique<cricket::FakeMediaEngine>()),
@@ -119,16 +119,16 @@
// Fake media channels are owned by the media engine.
voice_media_send_channel_ = media_engine_->voice().CreateSendChannel(
&fake_call_, cricket::MediaConfig(), cricket::AudioOptions(),
- webrtc::CryptoOptions(), webrtc::AudioCodecPairId::Create());
+ CryptoOptions(), AudioCodecPairId::Create());
video_media_send_channel_ = media_engine_->video().CreateSendChannel(
&fake_call_, cricket::MediaConfig(), cricket::VideoOptions(),
- webrtc::CryptoOptions(), video_bitrate_allocator_factory_.get());
+ CryptoOptions(), video_bitrate_allocator_factory_.get());
voice_media_receive_channel_ = media_engine_->voice().CreateReceiveChannel(
&fake_call_, cricket::MediaConfig(), cricket::AudioOptions(),
- webrtc::CryptoOptions(), webrtc::AudioCodecPairId::Create());
+ CryptoOptions(), AudioCodecPairId::Create());
video_media_receive_channel_ = media_engine_->video().CreateReceiveChannel(
&fake_call_, cricket::MediaConfig(), cricket::VideoOptions(),
- webrtc::CryptoOptions());
+ CryptoOptions());
// Create streams for predefined SSRCs. Streams need to exist in order
// for the senders and receievers to apply parameters to them.
@@ -162,8 +162,8 @@
audio_track_ = nullptr;
}
- std::unique_ptr<webrtc::RtpTransportInternal> CreateDtlsSrtpTransport() {
- auto dtls_srtp_transport = std::make_unique<webrtc::DtlsSrtpTransport>(
+ std::unique_ptr<RtpTransportInternal> CreateDtlsSrtpTransport() {
+ auto dtls_srtp_transport = std::make_unique<DtlsSrtpTransport>(
/*rtcp_mux_required=*/true, field_trials_);
dtls_srtp_transport->SetDtlsTransports(rtp_dtls_transport_.get(),
/*rtcp_dtls_transport=*/nullptr);
@@ -515,12 +515,12 @@
test::RunLoop run_loop_;
rtc::Thread* const network_thread_;
rtc::Thread* const worker_thread_;
- webrtc::RtcEventLogNull event_log_;
+ RtcEventLogNull event_log_;
// The `rtp_dtls_transport_` and `rtp_transport_` should be destroyed after
// the `channel_manager`.
std::unique_ptr<cricket::DtlsTransportInternal> rtp_dtls_transport_;
- std::unique_ptr<webrtc::RtpTransportInternal> rtp_transport_;
- std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>
+ std::unique_ptr<RtpTransportInternal> rtp_transport_;
+ std::unique_ptr<VideoBitrateAllocatorFactory>
video_bitrate_allocator_factory_;
std::unique_ptr<cricket::FakeMediaEngine> media_engine_;
rtc::UniqueRandomIdGenerator ssrc_generator_;
@@ -540,7 +540,7 @@
rtc::scoped_refptr<MediaStreamInterface> local_stream_;
rtc::scoped_refptr<VideoTrackInterface> video_track_;
rtc::scoped_refptr<AudioTrackInterface> audio_track_;
- webrtc::test::ScopedKeyValueConfig field_trials_;
+ test::ScopedKeyValueConfig field_trials_;
};
// Test that `voice_channel_` is updated when an audio track is associated
@@ -651,15 +651,13 @@
TEST_F(RtpSenderReceiverTest, RemoteVideoTrackState) {
CreateVideoRtpReceiver();
- EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, video_track_->state());
- EXPECT_EQ(webrtc::MediaSourceInterface::kLive,
- video_track_->GetSource()->state());
+ EXPECT_EQ(MediaStreamTrackInterface::kLive, video_track_->state());
+ EXPECT_EQ(MediaSourceInterface::kLive, video_track_->GetSource()->state());
DestroyVideoRtpReceiver();
- EXPECT_EQ(webrtc::MediaStreamTrackInterface::kEnded, video_track_->state());
- EXPECT_EQ(webrtc::MediaSourceInterface::kEnded,
- video_track_->GetSource()->state());
+ EXPECT_EQ(MediaStreamTrackInterface::kEnded, video_track_->state());
+ EXPECT_EQ(MediaSourceInterface::kEnded, video_track_->GetSource()->state());
DestroyVideoRtpReceiver();
}
@@ -888,9 +886,9 @@
RtpParameters params = audio_rtp_sender_->GetParameters();
EXPECT_EQ(1u, params.encodings.size());
- absl::optional<webrtc::RTCError> result;
+ absl::optional<RTCError> result;
audio_rtp_sender_->SetParametersAsync(
- params, [&result](webrtc::RTCError error) { result = error; });
+ params, [&result](RTCError error) { result = error; });
run_loop_.Flush();
EXPECT_TRUE(result->ok());
@@ -918,13 +916,13 @@
audio_rtp_sender_ =
AudioRtpSender::Create(worker_thread_, /*id=*/"", nullptr, nullptr);
- absl::optional<webrtc::RTCError> result;
+ absl::optional<RTCError> result;
RtpParameters params = audio_rtp_sender_->GetParameters();
ASSERT_EQ(1u, params.encodings.size());
params.encodings[0].max_bitrate_bps = 90000;
audio_rtp_sender_->SetParametersAsync(
- params, [&result](webrtc::RTCError error) { result = error; });
+ params, [&result](RTCError error) { result = error; });
run_loop_.Flush();
EXPECT_TRUE(result->ok());
@@ -932,7 +930,7 @@
EXPECT_EQ(params.encodings[0].max_bitrate_bps, 90000);
audio_rtp_sender_->SetParametersAsync(
- params, [&result](webrtc::RTCError error) { result = error; });
+ params, [&result](RTCError error) { result = error; });
run_loop_.Flush();
EXPECT_TRUE(result->ok());
@@ -1016,13 +1014,13 @@
RtpParameters params = audio_rtp_sender_->GetParameters();
EXPECT_EQ(1u, params.encodings.size());
- absl::optional<webrtc::RTCError> result;
+ absl::optional<RTCError> result;
audio_rtp_sender_->SetParametersAsync(
- params, [&result](webrtc::RTCError error) { result = error; });
+ params, [&result](RTCError error) { result = error; });
run_loop_.Flush();
EXPECT_TRUE(result->ok());
audio_rtp_sender_->SetParametersAsync(
- params, [&result](webrtc::RTCError error) { result = error; });
+ params, [&result](RTCError error) { result = error; });
run_loop_.Flush();
EXPECT_EQ(RTCErrorType::INVALID_STATE, result->type());
@@ -1081,7 +1079,7 @@
CreateAudioRtpSender();
EXPECT_EQ(-1, voice_media_send_channel()->max_bps());
- webrtc::RtpParameters params = audio_rtp_sender_->GetParameters();
+ RtpParameters params = audio_rtp_sender_->GetParameters();
EXPECT_EQ(1U, params.encodings.size());
EXPECT_FALSE(params.encodings[0].max_bitrate_bps);
params.encodings[0].max_bitrate_bps = 1000;
@@ -1106,10 +1104,9 @@
TEST_F(RtpSenderReceiverTest, SetAudioBitratePriority) {
CreateAudioRtpSender();
- webrtc::RtpParameters params = audio_rtp_sender_->GetParameters();
+ RtpParameters params = audio_rtp_sender_->GetParameters();
EXPECT_EQ(1U, params.encodings.size());
- EXPECT_EQ(webrtc::kDefaultBitratePriority,
- params.encodings[0].bitrate_priority);
+ EXPECT_EQ(kDefaultBitratePriority, params.encodings[0].bitrate_priority);
double new_bitrate_priority = 2.0;
params.encodings[0].bitrate_priority = new_bitrate_priority;
EXPECT_TRUE(audio_rtp_sender_->SetParameters(params).ok());
@@ -1140,9 +1137,9 @@
RtpParameters params = video_rtp_sender_->GetParameters();
EXPECT_EQ(1u, params.encodings.size());
- absl::optional<webrtc::RTCError> result;
+ absl::optional<RTCError> result;
video_rtp_sender_->SetParametersAsync(
- params, [&result](webrtc::RTCError error) { result = error; });
+ params, [&result](RTCError error) { result = error; });
run_loop_.Flush();
EXPECT_TRUE(result->ok());
@@ -1170,19 +1167,19 @@
video_rtp_sender_ =
VideoRtpSender::Create(worker_thread_, /*id=*/"", nullptr);
- absl::optional<webrtc::RTCError> result;
+ absl::optional<RTCError> result;
RtpParameters params = video_rtp_sender_->GetParameters();
ASSERT_EQ(1u, params.encodings.size());
params.encodings[0].max_bitrate_bps = 90000;
video_rtp_sender_->SetParametersAsync(
- params, [&result](webrtc::RTCError error) { result = error; });
+ params, [&result](RTCError error) { result = error; });
run_loop_.Flush();
EXPECT_TRUE(result->ok());
params = video_rtp_sender_->GetParameters();
EXPECT_EQ(params.encodings[0].max_bitrate_bps, 90000);
video_rtp_sender_->SetParametersAsync(
- params, [&result](webrtc::RTCError error) { result = error; });
+ params, [&result](RTCError error) { result = error; });
run_loop_.Flush();
EXPECT_TRUE(result->ok());
@@ -1350,13 +1347,13 @@
RtpParameters params = video_rtp_sender_->GetParameters();
EXPECT_EQ(1u, params.encodings.size());
- absl::optional<webrtc::RTCError> result;
+ absl::optional<RTCError> result;
video_rtp_sender_->SetParametersAsync(
- params, [&result](webrtc::RTCError error) { result = error; });
+ params, [&result](RTCError error) { result = error; });
run_loop_.Flush();
EXPECT_TRUE(result->ok());
video_rtp_sender_->SetParametersAsync(
- params, [&result](webrtc::RTCError error) { result = error; });
+ params, [&result](RTCError error) { result = error; });
run_loop_.Flush();
EXPECT_EQ(RTCErrorType::INVALID_STATE, result->type());
@@ -1453,7 +1450,7 @@
CreateVideoRtpSender();
RtpParameters params = video_rtp_sender_->GetParameters();
- params.encodings[0].num_temporal_layers = webrtc::kMaxTemporalStreams + 1;
+ params.encodings[0].num_temporal_layers = kMaxTemporalStreams + 1;
RTCError result = video_rtp_sender_->SetParameters(params);
EXPECT_EQ(RTCErrorType::INVALID_RANGE, result.type());
@@ -1536,7 +1533,7 @@
CreateVideoRtpSender();
EXPECT_EQ(-1, video_media_send_channel()->max_bps());
- webrtc::RtpParameters params = video_rtp_sender_->GetParameters();
+ RtpParameters params = video_rtp_sender_->GetParameters();
EXPECT_EQ(1U, params.encodings.size());
EXPECT_FALSE(params.encodings[0].min_bitrate_bps);
EXPECT_FALSE(params.encodings[0].max_bitrate_bps);
@@ -1589,10 +1586,9 @@
TEST_F(RtpSenderReceiverTest, SetVideoBitratePriority) {
CreateVideoRtpSender();
- webrtc::RtpParameters params = video_rtp_sender_->GetParameters();
+ RtpParameters params = video_rtp_sender_->GetParameters();
EXPECT_EQ(1U, params.encodings.size());
- EXPECT_EQ(webrtc::kDefaultBitratePriority,
- params.encodings[0].bitrate_priority);
+ EXPECT_EQ(kDefaultBitratePriority, params.encodings[0].bitrate_priority);
double new_bitrate_priority = 2.0;
params.encodings[0].bitrate_priority = new_bitrate_priority;
EXPECT_TRUE(video_rtp_sender_->SetParameters(params).ok());
diff --git a/pc/rtp_transceiver.cc b/pc/rtp_transceiver.cc
index 815ec9d..ca626cc9 100644
--- a/pc/rtp_transceiver.cc
+++ b/pc/rtp_transceiver.cc
@@ -542,7 +542,7 @@
RtpTransceiverDirection RtpTransceiver::direction() const {
if (unified_plan_ && stopping())
- return webrtc::RtpTransceiverDirection::kStopped;
+ return RtpTransceiverDirection::kStopped;
return direction_;
}
@@ -570,7 +570,7 @@
absl::optional<RtpTransceiverDirection> RtpTransceiver::current_direction()
const {
if (unified_plan_ && stopped())
- return webrtc::RtpTransceiverDirection::kStopped;
+ return RtpTransceiverDirection::kStopped;
return current_direction_;
}
@@ -604,7 +604,7 @@
});
stopping_ = true;
- direction_ = webrtc::RtpTransceiverDirection::kInactive;
+ direction_ = RtpTransceiverDirection::kInactive;
}
RTCError RtpTransceiver::StopStandard() {
diff --git a/pc/rtp_transceiver.h b/pc/rtp_transceiver.h
index deda5d7..88febb9 100644
--- a/pc/rtp_transceiver.h
+++ b/pc/rtp_transceiver.h
@@ -358,20 +358,18 @@
PROXY_CONSTMETHOD0(bool, stopped)
PROXY_CONSTMETHOD0(bool, stopping)
PROXY_CONSTMETHOD0(RtpTransceiverDirection, direction)
-PROXY_METHOD1(webrtc::RTCError, SetDirectionWithError, RtpTransceiverDirection)
+PROXY_METHOD1(RTCError, SetDirectionWithError, RtpTransceiverDirection)
PROXY_CONSTMETHOD0(absl::optional<RtpTransceiverDirection>, current_direction)
PROXY_CONSTMETHOD0(absl::optional<RtpTransceiverDirection>, fired_direction)
-PROXY_METHOD0(webrtc::RTCError, StopStandard)
+PROXY_METHOD0(RTCError, StopStandard)
PROXY_METHOD0(void, StopInternal)
-PROXY_METHOD1(webrtc::RTCError,
- SetCodecPreferences,
- rtc::ArrayView<RtpCodecCapability>)
+PROXY_METHOD1(RTCError, SetCodecPreferences, rtc::ArrayView<RtpCodecCapability>)
PROXY_CONSTMETHOD0(std::vector<RtpCodecCapability>, codec_preferences)
PROXY_CONSTMETHOD0(std::vector<RtpHeaderExtensionCapability>,
GetHeaderExtensionsToNegotiate)
PROXY_CONSTMETHOD0(std::vector<RtpHeaderExtensionCapability>,
GetNegotiatedHeaderExtensions)
-PROXY_METHOD1(webrtc::RTCError,
+PROXY_METHOD1(RTCError,
SetHeaderExtensionsToNegotiate,
rtc::ArrayView<const RtpHeaderExtensionCapability>)
END_PROXY_MAP(RtpTransceiver)
diff --git a/pc/rtp_transceiver_unittest.cc b/pc/rtp_transceiver_unittest.cc
index 63e06be..bd711f1 100644
--- a/pc/rtp_transceiver_unittest.cc
+++ b/pc/rtp_transceiver_unittest.cc
@@ -420,8 +420,8 @@
EXPECT_CALL(*mock_channel, mid()).WillRepeatedly(ReturnRef(content_name));
EXPECT_CALL(*mock_channel, SetRtpTransport(_)).WillRepeatedly(Return(true));
- cricket::RtpHeaderExtensions extensions = {webrtc::RtpExtension("uri1", 1),
- webrtc::RtpExtension("uri2", 2)};
+ cricket::RtpHeaderExtensions extensions = {RtpExtension("uri1", 1),
+ RtpExtension("uri2", 2)};
cricket::AudioContentDescription description;
description.set_rtp_header_extensions(extensions);
transceiver_->OnNegotiationUpdate(SdpType::kAnswer, &description);
@@ -449,8 +449,8 @@
EXPECT_CALL(*sender_.get(), SetTransceiverAsStopped());
EXPECT_CALL(*sender_.get(), Stop());
- cricket::RtpHeaderExtensions extensions = {webrtc::RtpExtension("uri1", 1),
- webrtc::RtpExtension("uri2", 2)};
+ cricket::RtpHeaderExtensions extensions = {RtpExtension("uri1", 1),
+ RtpExtension("uri2", 2)};
cricket::AudioContentDescription description;
description.set_rtp_header_extensions(extensions);
transceiver_->OnNegotiationUpdate(SdpType::kAnswer, &description);
@@ -464,8 +464,7 @@
RtpTransceiverDirection::kStopped),
Field(&RtpHeaderExtensionCapability::direction,
RtpTransceiverDirection::kStopped)));
- extensions = {webrtc::RtpExtension("uri3", 4),
- webrtc::RtpExtension("uri5", 6)};
+ extensions = {RtpExtension("uri3", 4), RtpExtension("uri5", 6)};
description.set_rtp_header_extensions(extensions);
transceiver_->OnNegotiationUpdate(SdpType::kAnswer, &description);
diff --git a/pc/rtp_transport.cc b/pc/rtp_transport.cc
index 2ffb53f..7cf9fe0 100644
--- a/pc/rtp_transport.cc
+++ b/pc/rtp_transport.cc
@@ -186,10 +186,10 @@
void RtpTransport::DemuxPacket(rtc::CopyOnWriteBuffer packet,
int64_t packet_time_us) {
- webrtc::RtpPacketReceived parsed_packet(
- &header_extension_map_, packet_time_us == -1
- ? Timestamp::MinusInfinity()
- : Timestamp::Micros(packet_time_us));
+ RtpPacketReceived parsed_packet(&header_extension_map_,
+ packet_time_us == -1
+ ? Timestamp::MinusInfinity()
+ : Timestamp::Micros(packet_time_us));
if (!parsed_packet.Parse(std::move(packet))) {
RTC_LOG(LS_ERROR)
<< "Failed to parse the incoming RTP packet before demuxing. Drop it.";
diff --git a/pc/rtp_transport_internal.h b/pc/rtp_transport_internal.h
index 4114fa9..483a1ce 100644
--- a/pc/rtp_transport_internal.h
+++ b/pc/rtp_transport_internal.h
@@ -72,7 +72,7 @@
// Called whenever a RTP packet that can not be demuxed by the transport is
// received.
void SetUnDemuxableRtpPacketReceivedHandler(
- absl::AnyInvocable<void(webrtc::RtpPacketReceived&)> callback) {
+ absl::AnyInvocable<void(RtpPacketReceived&)> callback) {
callback_undemuxable_rtp_packet_received_ = std::move(callback);
}
@@ -160,7 +160,7 @@
CallbackList<bool> callback_list_ready_to_send_;
CallbackList<rtc::CopyOnWriteBuffer*, int64_t>
callback_list_rtcp_packet_received_;
- absl::AnyInvocable<void(webrtc::RtpPacketReceived&)>
+ absl::AnyInvocable<void(RtpPacketReceived&)>
callback_undemuxable_rtp_packet_received_ =
[](RtpPacketReceived& packet) {};
CallbackList<absl::optional<rtc::NetworkRoute>>
diff --git a/pc/sctp_transport.h b/pc/sctp_transport.h
index 35e7656..076dee5 100644
--- a/pc/sctp_transport.h
+++ b/pc/sctp_transport.h
@@ -61,7 +61,7 @@
void Start(int local_port, int remote_port, int max_message_size);
// TODO(https://bugs.webrtc.org/10629): Move functions that need
- // internal() to be functions on the webrtc::SctpTransport interface,
+ // internal() to be functions on the SctpTransport interface,
// and make the internal() function private.
cricket::SctpTransportInternal* internal() {
RTC_DCHECK_RUN_ON(owner_thread_);
diff --git a/pc/sdp_offer_answer.cc b/pc/sdp_offer_answer.cc
index 0261195..04d1aff 100644
--- a/pc/sdp_offer_answer.cc
+++ b/pc/sdp_offer_answer.cc
@@ -86,8 +86,7 @@
namespace {
-typedef webrtc::PeerConnectionInterface::RTCOfferAnswerOptions
- RTCOfferAnswerOptions;
+typedef PeerConnectionInterface::RTCOfferAnswerOptions RTCOfferAnswerOptions;
// Error messages
const char kInvalidSdp[] = "Invalid session description.";
@@ -834,8 +833,8 @@
}
// Check if we can send `new_stream` on a PeerConnection.
-bool CanAddLocalMediaStream(webrtc::StreamCollectionInterface* current_streams,
- webrtc::MediaStreamInterface* new_stream) {
+bool CanAddLocalMediaStream(StreamCollectionInterface* current_streams,
+ MediaStreamInterface* new_stream) {
if (!new_stream || !current_streams) {
return false;
}
@@ -847,7 +846,7 @@
return true;
}
-rtc::scoped_refptr<webrtc::DtlsTransport> LookupDtlsTransportByMid(
+rtc::scoped_refptr<DtlsTransport> LookupDtlsTransportByMid(
rtc::Thread* network_thread,
JsepTransportController* controller,
const std::string& mid) {
diff --git a/pc/sdp_offer_answer.h b/pc/sdp_offer_answer.h
index 8aa7040..88ddfe0 100644
--- a/pc/sdp_offer_answer.h
+++ b/pc/sdp_offer_answer.h
@@ -674,8 +674,8 @@
// or else the CreateBuiltinVideoBitrateAllocatorFactory() will be called.
// Note that one can still choose to override this in a MediaEngine
// if one wants too.
- std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>
- video_bitrate_allocator_factory_ RTC_GUARDED_BY(signaling_thread());
+ std::unique_ptr<VideoBitrateAllocatorFactory> video_bitrate_allocator_factory_
+ RTC_GUARDED_BY(signaling_thread());
// Whether we are the initial offerer on the association. This
// determines the SSL role.
diff --git a/pc/sdp_offer_answer_unittest.cc b/pc/sdp_offer_answer_unittest.cc
index 94ceff1..9a44360 100644
--- a/pc/sdp_offer_answer_unittest.cc
+++ b/pc/sdp_offer_answer_unittest.cc
@@ -88,7 +88,7 @@
Dav1dDecoderTemplateAdapter>>(),
nullptr /* audio_mixer */,
nullptr /* audio_processing */)) {
- webrtc::metrics::Reset();
+ metrics::Reset();
}
std::unique_ptr<PeerConnectionWrapper> CreatePeerConnection() {
@@ -168,8 +168,8 @@
// There is no error yet but the metrics counter will increase.
EXPECT_TRUE(error.ok());
EXPECT_METRIC_EQ(
- 1, webrtc::metrics::NumEvents(
- "WebRTC.PeerConnection.ValidBundledPayloadTypes", false));
+ 1, metrics::NumEvents("WebRTC.PeerConnection.ValidBundledPayloadTypes",
+ false));
// Tolerate codec collisions in rejected m-lines.
pc = CreatePeerConnection();
@@ -178,9 +178,9 @@
absl::StrReplaceAll(sdp, {{"m=video 9 ", "m=video 0 "}}));
pc->SetRemoteDescription(std::move(rejected_offer), &error);
EXPECT_TRUE(error.ok());
- EXPECT_METRIC_EQ(1,
- webrtc::metrics::NumEvents(
- "WebRTC.PeerConnection.ValidBundledPayloadTypes", true));
+ EXPECT_METRIC_EQ(
+ 1, metrics::NumEvents("WebRTC.PeerConnection.ValidBundledPayloadTypes",
+ true));
}
TEST_F(SdpOfferAnswerTest, BundleRejectsCodecCollisionsVideoFmtp) {
@@ -221,8 +221,8 @@
pc->SetRemoteDescription(std::move(desc), &error);
EXPECT_TRUE(error.ok());
EXPECT_METRIC_EQ(
- 1, webrtc::metrics::NumEvents(
- "WebRTC.PeerConnection.ValidBundledPayloadTypes", false));
+ 1, metrics::NumEvents("WebRTC.PeerConnection.ValidBundledPayloadTypes",
+ false));
}
TEST_F(SdpOfferAnswerTest, BundleCodecCollisionInDifferentBundlesAllowed) {
@@ -264,8 +264,8 @@
pc->SetRemoteDescription(std::move(desc), &error);
EXPECT_TRUE(error.ok());
EXPECT_METRIC_EQ(
- 0, webrtc::metrics::NumEvents(
- "WebRTC.PeerConnection.ValidBundledPayloadTypes", false));
+ 0, metrics::NumEvents("WebRTC.PeerConnection.ValidBundledPayloadTypes",
+ false));
}
TEST_F(SdpOfferAnswerTest, BundleMeasuresHeaderExtensionIdCollision) {
diff --git a/pc/slow_peer_connection_integration_test.cc b/pc/slow_peer_connection_integration_test.cc
index fd9d341..4e26283 100644
--- a/pc/slow_peer_connection_integration_test.cc
+++ b/pc/slow_peer_connection_integration_test.cc
@@ -67,7 +67,7 @@
// Some things use a time of "0" as a special value, so we need to start out
// the fake clock at a nonzero time.
// TODO(deadbeef): Fix this.
- AdvanceTime(webrtc::TimeDelta::Seconds(1000));
+ AdvanceTime(TimeDelta::Seconds(1000));
}
// Explicit handle.
@@ -170,20 +170,20 @@
CreateTurnServer(turn_server_internal_address, turn_server_external_address,
cricket::PROTO_TLS, "88.88.88.0");
- webrtc::PeerConnectionInterface::IceServer ice_server;
+ PeerConnectionInterface::IceServer ice_server;
ice_server.urls.push_back("turns:88.88.88.0:3478?transport=tcp");
ice_server.username = "test";
ice_server.password = "test";
PeerConnectionInterface::RTCConfiguration client_1_config;
client_1_config.servers.push_back(ice_server);
- client_1_config.type = webrtc::PeerConnectionInterface::kRelay;
+ client_1_config.type = PeerConnectionInterface::kRelay;
PeerConnectionInterface::RTCConfiguration client_2_config;
client_2_config.servers.push_back(ice_server);
// Setting the type to kRelay forces the connection to go through a TURN
// server.
- client_2_config.type = webrtc::PeerConnectionInterface::kRelay;
+ client_2_config.type = PeerConnectionInterface::kRelay;
// Get a copy to the pointer so we can verify calls later.
rtc::TestCertificateVerifier* client_1_cert_verifier =
@@ -194,10 +194,10 @@
client_2_cert_verifier->verify_certificate_ = false;
// Create the dependencies with the test certificate verifier.
- webrtc::PeerConnectionDependencies client_1_deps(nullptr);
+ PeerConnectionDependencies client_1_deps(nullptr);
client_1_deps.tls_cert_verifier =
std::unique_ptr<rtc::TestCertificateVerifier>(client_1_cert_verifier);
- webrtc::PeerConnectionDependencies client_2_deps(nullptr);
+ PeerConnectionDependencies client_2_deps(nullptr);
client_2_deps.tls_cert_verifier =
std::unique_ptr<rtc::TestCertificateVerifier>(client_2_cert_verifier);
diff --git a/pc/srtp_transport_unittest.cc b/pc/srtp_transport_unittest.cc
index fead5d6..de4ff03 100644
--- a/pc/srtp_transport_unittest.cc
+++ b/pc/srtp_transport_unittest.cc
@@ -342,7 +342,7 @@
TransportObserver rtp_sink2_;
int sequence_number_ = 0;
- webrtc::test::ScopedKeyValueConfig field_trials_;
+ test::ScopedKeyValueConfig field_trials_;
};
class SrtpTransportTestWithExternalAuth
diff --git a/pc/test/android_test_initializer.cc b/pc/test/android_test_initializer.cc
index 963544c..88b4587 100644
--- a/pc/test/android_test_initializer.cc
+++ b/pc/test/android_test_initializer.cc
@@ -39,7 +39,7 @@
RTC_CHECK(rtc::InitializeSSL()) << "Failed to InitializeSSL()";
- webrtc::JVM::Initialize(jvm);
+ JVM::Initialize(jvm);
}
} // anonymous namespace
diff --git a/pc/test/fake_peer_connection_base.h b/pc/test/fake_peer_connection_base.h
index a1c8dca..1615088 100644
--- a/pc/test/fake_peer_connection_base.h
+++ b/pc/test/fake_peer_connection_base.h
@@ -363,7 +363,7 @@
const FieldTrialsView& trials() const override { return field_trials_; }
protected:
- webrtc::test::ScopedKeyValueConfig field_trials_;
+ test::ScopedKeyValueConfig field_trials_;
};
} // namespace webrtc
diff --git a/pc/test/fake_peer_connection_for_stats.h b/pc/test/fake_peer_connection_for_stats.h
index a65e6d5..33a5361 100644
--- a/pc/test/fake_peer_connection_for_stats.h
+++ b/pc/test/fake_peer_connection_for_stats.h
@@ -150,7 +150,7 @@
receive_channel,
const std::string& content_name,
bool srtp_required,
- webrtc::CryptoOptions crypto_options,
+ CryptoOptions crypto_options,
rtc::UniqueRandomIdGenerator* ssrc_generator,
std::string transport_name)
: VoiceChannel(worker_thread,
@@ -183,7 +183,7 @@
receive_channel,
const std::string& content_name,
bool srtp_required,
- webrtc::CryptoOptions crypto_options,
+ CryptoOptions crypto_options,
rtc::UniqueRandomIdGenerator* ssrc_generator,
std::string transport_name)
: VideoChannel(worker_thread,
@@ -298,7 +298,7 @@
worker_thread_, network_thread_, signaling_thread_,
std::move(voice_media_send_channel),
std::move(voice_media_receive_channel), mid, kDefaultSrtpRequired,
- webrtc::CryptoOptions(), context_->ssrc_generator(), transport_name);
+ CryptoOptions(), context_->ssrc_generator(), transport_name);
auto transceiver =
GetOrCreateFirstTransceiverOfType(cricket::MEDIA_TYPE_AUDIO)
->internal();
@@ -332,7 +332,7 @@
worker_thread_, network_thread_, signaling_thread_,
std::move(video_media_send_channel),
std::move(video_media_receive_channel), mid, kDefaultSrtpRequired,
- webrtc::CryptoOptions(), context_->ssrc_generator(), transport_name);
+ CryptoOptions(), context_->ssrc_generator(), transport_name);
auto transceiver =
GetOrCreateFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO)
->internal();
diff --git a/pc/test/fake_periodic_video_source.h b/pc/test/fake_periodic_video_source.h
index 452a8f6..65652bd 100644
--- a/pc/test/fake_periodic_video_source.h
+++ b/pc/test/fake_periodic_video_source.h
@@ -65,12 +65,12 @@
return wants_;
}
- void RemoveSink(rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) override {
+ void RemoveSink(rtc::VideoSinkInterface<VideoFrame>* sink) override {
RTC_DCHECK(thread_checker_.IsCurrent());
broadcaster_.RemoveSink(sink);
}
- void AddOrUpdateSink(rtc::VideoSinkInterface<webrtc::VideoFrame>* sink,
+ void AddOrUpdateSink(rtc::VideoSinkInterface<VideoFrame>* sink,
const rtc::VideoSinkWants& wants) override {
RTC_DCHECK(thread_checker_.IsCurrent());
{
diff --git a/pc/test/integration_test_helpers.cc b/pc/test/integration_test_helpers.cc
index ede159d..64d8deb 100644
--- a/pc/test/integration_test_helpers.cc
+++ b/pc/test/integration_test_helpers.cc
@@ -46,7 +46,7 @@
int FindFirstMediaStatsIndexByKind(
const std::string& kind,
- const std::vector<const webrtc::RTCInboundRtpStreamStats*>& inbound_rtps) {
+ const std::vector<const RTCInboundRtpStreamStats*>& inbound_rtps) {
for (size_t i = 0; i < inbound_rtps.size(); i++) {
if (*inbound_rtps[i]->kind == kind) {
return i;
diff --git a/pc/test/integration_test_helpers.h b/pc/test/integration_test_helpers.h
index c61712c..1ba3326 100644
--- a/pc/test/integration_test_helpers.h
+++ b/pc/test/integration_test_helpers.h
@@ -177,14 +177,14 @@
int FindFirstMediaStatsIndexByKind(
const std::string& kind,
- const std::vector<const webrtc::RTCInboundRtpStreamStats*>& inbound_rtps);
+ const std::vector<const RTCInboundRtpStreamStats*>& inbound_rtps);
-class TaskQueueMetronome : public webrtc::Metronome {
+class TaskQueueMetronome : public Metronome {
public:
explicit TaskQueueMetronome(TimeDelta tick_period);
~TaskQueueMetronome() override;
- // webrtc::Metronome implementation.
+ // Metronome implementation.
void RequestCallOnNextTick(absl::AnyInvocable<void() &&> callback) override;
TimeDelta TickPeriod() const override;
@@ -207,7 +207,7 @@
virtual ~SignalingMessageReceiver() {}
};
-class MockRtpReceiverObserver : public webrtc::RtpReceiverObserverInterface {
+class MockRtpReceiverObserver : public RtpReceiverObserverInterface {
public:
explicit MockRtpReceiverObserver(cricket::MediaType media_type)
: expected_media_type_(media_type) {}
@@ -234,14 +234,14 @@
// advertise support of any codecs.
// TODO(steveanton): See how this could become a subclass of
// PeerConnectionWrapper defined in peerconnectionwrapper.h.
-class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
+class PeerConnectionIntegrationWrapper : public PeerConnectionObserver,
public SignalingMessageReceiver {
public:
- webrtc::PeerConnectionFactoryInterface* pc_factory() const {
+ PeerConnectionFactoryInterface* pc_factory() const {
return peer_connection_factory_.get();
}
- webrtc::PeerConnectionInterface* pc() const { return peer_connection_.get(); }
+ PeerConnectionInterface* pc() const { return peer_connection_.get(); }
// If a signaling message receiver is set (via ConnectFakeSignaling), this
// will set the whole offer/answer exchange in motion. Just need to wait for
@@ -339,11 +339,11 @@
return AddTrack(CreateLocalVideoTrack());
}
- rtc::scoped_refptr<webrtc::AudioTrackInterface> CreateLocalAudioTrack() {
+ rtc::scoped_refptr<AudioTrackInterface> CreateLocalAudioTrack() {
cricket::AudioOptions options;
// Disable highpass filter so that we can get all the test audio frames.
options.highpass_filter = false;
- rtc::scoped_refptr<webrtc::AudioSourceInterface> source =
+ rtc::scoped_refptr<AudioSourceInterface> source =
peer_connection_factory_->CreateAudioSource(options);
// TODO(perkj): Test audio source when it is implemented. Currently audio
// always use the default input.
@@ -351,21 +351,20 @@
source.get());
}
- rtc::scoped_refptr<webrtc::VideoTrackInterface> CreateLocalVideoTrack() {
- webrtc::FakePeriodicVideoSource::Config config;
+ rtc::scoped_refptr<VideoTrackInterface> CreateLocalVideoTrack() {
+ FakePeriodicVideoSource::Config config;
config.timestamp_offset_ms = rtc::TimeMillis();
return CreateLocalVideoTrackInternal(config);
}
- rtc::scoped_refptr<webrtc::VideoTrackInterface>
- CreateLocalVideoTrackWithConfig(
- webrtc::FakePeriodicVideoSource::Config config) {
+ rtc::scoped_refptr<VideoTrackInterface> CreateLocalVideoTrackWithConfig(
+ FakePeriodicVideoSource::Config config) {
return CreateLocalVideoTrackInternal(config);
}
- rtc::scoped_refptr<webrtc::VideoTrackInterface>
- CreateLocalVideoTrackWithRotation(webrtc::VideoRotation rotation) {
- webrtc::FakePeriodicVideoSource::Config config;
+ rtc::scoped_refptr<VideoTrackInterface> CreateLocalVideoTrackWithRotation(
+ VideoRotation rotation) {
+ FakePeriodicVideoSource::Config config;
config.rotation = rotation;
config.timestamp_offset_ms = rtc::TimeMillis();
return CreateLocalVideoTrackInternal(config);
@@ -409,22 +408,22 @@
}
bool SignalingStateStable() {
- return pc()->signaling_state() == webrtc::PeerConnectionInterface::kStable;
+ return pc()->signaling_state() == PeerConnectionInterface::kStable;
}
bool IceGatheringStateComplete() {
return pc()->ice_gathering_state() ==
- webrtc::PeerConnectionInterface::kIceGatheringComplete;
+ PeerConnectionInterface::kIceGatheringComplete;
}
void CreateDataChannel() { CreateDataChannel(nullptr); }
- void CreateDataChannel(const webrtc::DataChannelInit* init) {
+ void CreateDataChannel(const DataChannelInit* init) {
CreateDataChannel(kDataChannelLabel, init);
}
void CreateDataChannel(const std::string& label,
- const webrtc::DataChannelInit* init) {
+ const DataChannelInit* init) {
auto data_channel_or_error = pc()->CreateDataChannelOrError(label, init);
ASSERT_TRUE(data_channel_or_error.ok());
data_channels_.push_back(data_channel_or_error.MoveValue());
@@ -482,7 +481,7 @@
// Returns a MockStatsObserver in a state after stats gathering finished,
// which can be used to access the gathered stats.
rtc::scoped_refptr<MockStatsObserver> OldGetStatsForTrack(
- webrtc::MediaStreamTrackInterface* track) {
+ MediaStreamTrackInterface* track) {
auto observer = rtc::make_ref_counted<MockStatsObserver>();
EXPECT_TRUE(peer_connection_->GetStats(
observer.get(), nullptr,
@@ -498,9 +497,8 @@
// Synchronously gets stats and returns them. If it times out, fails the test
// and returns null.
- rtc::scoped_refptr<const webrtc::RTCStatsReport> NewGetStats() {
- auto callback =
- rtc::make_ref_counted<webrtc::MockRTCStatsCollectorCallback>();
+ rtc::scoped_refptr<const RTCStatsReport> NewGetStats() {
+ auto callback = rtc::make_ref_counted<MockRTCStatsCollectorCallback>();
peer_connection_->GetStats(callback.get());
EXPECT_TRUE_WAIT(callback->called(), kDefaultTimeout);
return callback->report();
@@ -527,10 +525,10 @@
return static_cast<double>(rendered_width()) / rendered_height();
}
- webrtc::VideoRotation rendered_rotation() {
+ VideoRotation rendered_rotation() {
EXPECT_FALSE(fake_video_renderers_.empty());
return fake_video_renderers_.empty()
- ? webrtc::kVideoRotation_0
+ ? kVideoRotation_0
: fake_video_renderers_.begin()->second->rotation();
}
@@ -573,20 +571,20 @@
return pc()->local_streams().get();
}
- webrtc::PeerConnectionInterface::SignalingState signaling_state() {
+ PeerConnectionInterface::SignalingState signaling_state() {
return pc()->signaling_state();
}
- webrtc::PeerConnectionInterface::IceConnectionState ice_connection_state() {
+ PeerConnectionInterface::IceConnectionState ice_connection_state() {
return pc()->ice_connection_state();
}
- webrtc::PeerConnectionInterface::IceConnectionState
+ PeerConnectionInterface::IceConnectionState
standardized_ice_connection_state() {
return pc()->standardized_ice_connection_state();
}
- webrtc::PeerConnectionInterface::IceGatheringState ice_gathering_state() {
+ PeerConnectionInterface::IceGatheringState ice_gathering_state() {
return pc()->ice_gathering_state();
}
@@ -615,7 +613,7 @@
}
cricket::PortAllocator* port_allocator() const { return port_allocator_; }
- webrtc::FakeRtcEventLogFactory* event_log_factory() const {
+ FakeRtcEventLogFactory* event_log_factory() const {
return event_log_factory_;
}
@@ -628,8 +626,7 @@
// Sets the mDNS responder for the owned fake network manager and keeps a
// reference to the responder.
- void SetMdnsResponder(
- std::unique_ptr<webrtc::FakeMdnsResponder> mdns_responder) {
+ void SetMdnsResponder(std::unique_ptr<FakeMdnsResponder> mdns_responder) {
RTC_DCHECK(mdns_responder != nullptr);
mdns_responder_ = mdns_responder.get();
network_manager()->set_mdns_responder(std::move(mdns_responder));
@@ -644,7 +641,7 @@
}
bool Rollback() {
return SetRemoteDescription(
- webrtc::CreateSessionDescription(SdpType::kRollback, ""));
+ CreateSessionDescription(SdpType::kRollback, ""));
}
// Functions for querying stats.
@@ -652,7 +649,7 @@
// Get the baseline numbers for audio_packets and audio_delay.
auto received_stats = NewGetStats();
auto rtp_stats =
- received_stats->GetStatsOfType<webrtc::RTCInboundRtpStreamStats>()[0];
+ received_stats->GetStatsOfType<RTCInboundRtpStreamStats>()[0];
ASSERT_TRUE(rtp_stats->relative_packet_arrival_delay.is_defined());
ASSERT_TRUE(rtp_stats->packets_received.is_defined());
rtp_stats_id_ = rtp_stats->id();
@@ -664,8 +661,7 @@
void UpdateDelayStats(std::string tag, int desc_size) {
auto report = NewGetStats();
- auto rtp_stats =
- report->GetAs<webrtc::RTCInboundRtpStreamStats>(rtp_stats_id_);
+ auto rtp_stats = report->GetAs<RTCInboundRtpStreamStats>(rtp_stats_id_);
ASSERT_TRUE(rtp_stats);
auto delta_packets = *rtp_stats->packets_received - audio_packets_stat_;
auto delta_rpad =
@@ -744,11 +740,11 @@
bool Init(const PeerConnectionFactory::Options* options,
const PeerConnectionInterface::RTCConfiguration* config,
- webrtc::PeerConnectionDependencies dependencies,
+ PeerConnectionDependencies dependencies,
rtc::SocketServer* socket_server,
rtc::Thread* network_thread,
rtc::Thread* worker_thread,
- std::unique_ptr<webrtc::FakeRtcEventLogFactory> event_log_factory,
+ std::unique_ptr<FakeRtcEventLogFactory> event_log_factory,
bool reset_encoder_factory,
bool reset_decoder_factory,
bool create_media_engine) {
@@ -771,12 +767,12 @@
}
rtc::Thread* const signaling_thread = rtc::Thread::Current();
- webrtc::PeerConnectionFactoryDependencies pc_factory_dependencies;
+ PeerConnectionFactoryDependencies pc_factory_dependencies;
pc_factory_dependencies.network_thread = network_thread;
pc_factory_dependencies.worker_thread = worker_thread;
pc_factory_dependencies.signaling_thread = signaling_thread;
pc_factory_dependencies.task_queue_factory =
- webrtc::CreateDefaultTaskQueueFactory();
+ CreateDefaultTaskQueueFactory();
pc_factory_dependencies.trials = std::make_unique<FieldTrialBasedConfig>();
pc_factory_dependencies.metronome =
std::make_unique<TaskQueueMetronome>(TimeDelta::Millis(8));
@@ -805,11 +801,11 @@
pc_factory_dependencies.event_log_factory = std::move(event_log_factory);
} else {
pc_factory_dependencies.event_log_factory =
- std::make_unique<webrtc::RtcEventLogFactory>(
+ std::make_unique<RtcEventLogFactory>(
pc_factory_dependencies.task_queue_factory.get());
}
- peer_connection_factory_ = webrtc::CreateModularPeerConnectionFactory(
- std::move(pc_factory_dependencies));
+ peer_connection_factory_ =
+ CreateModularPeerConnectionFactory(std::move(pc_factory_dependencies));
if (!peer_connection_factory_) {
return false;
@@ -826,9 +822,9 @@
return peer_connection_.get() != nullptr;
}
- rtc::scoped_refptr<webrtc::PeerConnectionInterface> CreatePeerConnection(
+ rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
const PeerConnectionInterface::RTCConfiguration* config,
- webrtc::PeerConnectionDependencies dependencies) {
+ PeerConnectionDependencies dependencies) {
PeerConnectionInterface::RTCConfiguration modified_config;
modified_config.sdp_semantics = sdp_semantics_;
// If `config` is null, this will result in a default configuration being
@@ -861,21 +857,20 @@
signal_ice_candidates_ = signal;
}
- rtc::scoped_refptr<webrtc::VideoTrackInterface> CreateLocalVideoTrackInternal(
- webrtc::FakePeriodicVideoSource::Config config) {
+ rtc::scoped_refptr<VideoTrackInterface> CreateLocalVideoTrackInternal(
+ FakePeriodicVideoSource::Config config) {
// Set max frame rate to 10fps to reduce the risk of test flakiness.
// TODO(deadbeef): Do something more robust.
config.frame_interval_ms = 100;
video_track_sources_.emplace_back(
- rtc::make_ref_counted<webrtc::FakePeriodicVideoTrackSource>(
+ rtc::make_ref_counted<FakePeriodicVideoTrackSource>(
config, false /* remote */));
- rtc::scoped_refptr<webrtc::VideoTrackInterface> track =
+ rtc::scoped_refptr<VideoTrackInterface> track =
peer_connection_factory_->CreateVideoTrack(video_track_sources_.back(),
rtc::CreateRandomUuid());
if (!local_video_renderer_) {
- local_video_renderer_.reset(
- new webrtc::FakeVideoTrackRenderer(track.get()));
+ local_video_renderer_.reset(new FakeVideoTrackRenderer(track.get()));
}
return track;
}
@@ -883,7 +878,7 @@
void HandleIncomingOffer(const std::string& msg) {
RTC_LOG(LS_INFO) << debug_name_ << ": HandleIncomingOffer";
std::unique_ptr<SessionDescriptionInterface> desc =
- webrtc::CreateSessionDescription(SdpType::kOffer, msg);
+ CreateSessionDescription(SdpType::kOffer, msg);
if (received_sdp_munger_) {
received_sdp_munger_(desc->description());
}
@@ -903,7 +898,7 @@
void HandleIncomingAnswer(const std::string& msg) {
RTC_LOG(LS_INFO) << debug_name_ << ": HandleIncomingAnswer";
std::unique_ptr<SessionDescriptionInterface> desc =
- webrtc::CreateSessionDescription(SdpType::kAnswer, msg);
+ CreateSessionDescription(SdpType::kAnswer, msg);
if (received_sdp_munger_) {
received_sdp_munger_(desc->description());
}
@@ -1054,7 +1049,7 @@
const std::string& msg) override {
RTC_LOG(LS_INFO) << debug_name_ << ": ReceiveIceMessage";
absl::optional<RTCError> result;
- pc()->AddIceCandidate(absl::WrapUnique(webrtc::CreateIceCandidate(
+ pc()->AddIceCandidate(absl::WrapUnique(CreateIceCandidate(
sdp_mid, sdp_mline_index, msg, nullptr)),
[&result](RTCError r) { result = r; });
EXPECT_TRUE_WAIT(result.has_value(), kDefaultTimeout);
@@ -1063,7 +1058,7 @@
// PeerConnectionObserver callbacks.
void OnSignalingChange(
- webrtc::PeerConnectionInterface::SignalingState new_state) override {
+ PeerConnectionInterface::SignalingState new_state) override {
EXPECT_EQ(pc()->signaling_state(), new_state);
peer_connection_signaling_state_history_.push_back(new_state);
}
@@ -1092,21 +1087,21 @@
}
void OnRenegotiationNeeded() override {}
void OnIceConnectionChange(
- webrtc::PeerConnectionInterface::IceConnectionState new_state) override {
+ PeerConnectionInterface::IceConnectionState new_state) override {
EXPECT_EQ(pc()->ice_connection_state(), new_state);
ice_connection_state_history_.push_back(new_state);
}
void OnStandardizedIceConnectionChange(
- webrtc::PeerConnectionInterface::IceConnectionState new_state) override {
+ PeerConnectionInterface::IceConnectionState new_state) override {
standardized_ice_connection_state_history_.push_back(new_state);
}
void OnConnectionChange(
- webrtc::PeerConnectionInterface::PeerConnectionState new_state) override {
+ PeerConnectionInterface::PeerConnectionState new_state) override {
peer_connection_state_history_.push_back(new_state);
}
void OnIceGatheringChange(
- webrtc::PeerConnectionInterface::IceGatheringState new_state) override {
+ PeerConnectionInterface::IceGatheringState new_state) override {
EXPECT_EQ(pc()->ice_gathering_state(), new_state);
ice_gathering_state_history_.push_back(new_state);
}
@@ -1116,7 +1111,7 @@
ice_candidate_pair_change_history_.push_back(event);
}
- void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override {
+ void OnIceCandidate(const IceCandidateInterface* candidate) override {
RTC_LOG(LS_INFO) << debug_name_ << ": OnIceCandidate";
if (remote_async_dns_resolver_) {
@@ -1172,20 +1167,19 @@
std::unique_ptr<rtc::FakeNetworkManager> fake_network_manager_;
std::unique_ptr<rtc::BasicPacketSocketFactory> socket_factory_;
// Reference to the mDNS responder owned by `fake_network_manager_` after set.
- webrtc::FakeMdnsResponder* mdns_responder_ = nullptr;
+ FakeMdnsResponder* mdns_responder_ = nullptr;
- rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_;
- rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface>
- peer_connection_factory_;
+ rtc::scoped_refptr<PeerConnectionInterface> peer_connection_;
+ rtc::scoped_refptr<PeerConnectionFactoryInterface> peer_connection_factory_;
cricket::PortAllocator* port_allocator_;
// Needed to keep track of number of frames sent.
rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
// Needed to keep track of number of frames received.
- std::map<std::string, std::unique_ptr<webrtc::FakeVideoTrackRenderer>>
+ std::map<std::string, std::unique_ptr<FakeVideoTrackRenderer>>
fake_video_renderers_;
// Needed to ensure frames aren't received for removed tracks.
- std::vector<std::unique_ptr<webrtc::FakeVideoTrackRenderer>>
+ std::vector<std::unique_ptr<FakeVideoTrackRenderer>>
removed_fake_video_renderers_;
// For remote peer communication.
@@ -1197,10 +1191,9 @@
// Store references to the video sources we've created, so that we can stop
// them, if required.
- std::vector<rtc::scoped_refptr<webrtc::VideoTrackSource>>
- video_track_sources_;
+ std::vector<rtc::scoped_refptr<VideoTrackSource>> video_track_sources_;
// `local_video_renderer_` attached to the first created local video track.
- std::unique_ptr<webrtc::FakeVideoTrackRenderer> local_video_renderer_;
+ std::unique_ptr<FakeVideoTrackRenderer> local_video_renderer_;
SdpSemantics sdp_semantics_;
PeerConnectionInterface::RTCOfferAnswerOptions offer_answer_options_;
@@ -1230,7 +1223,7 @@
ice_candidate_pair_change_history_;
std::vector<PeerConnectionInterface::SignalingState>
peer_connection_signaling_state_history_;
- webrtc::FakeRtcEventLogFactory* event_log_factory_;
+ FakeRtcEventLogFactory* event_log_factory_;
// Number of ICE candidates expected. The default is no limit.
int candidates_expected_ = std::numeric_limits<int>::max();
@@ -1247,7 +1240,7 @@
friend class PeerConnectionIntegrationBaseTest;
};
-class MockRtcEventLogOutput : public webrtc::RtcEventLogOutput {
+class MockRtcEventLogOutput : public RtcEventLogOutput {
public:
virtual ~MockRtcEventLogOutput() = default;
MOCK_METHOD(bool, IsActive, (), (const, override));
@@ -1359,7 +1352,7 @@
int callee_video_frames_expected_ = 0;
};
-class MockIceTransport : public webrtc::IceTransportInterface {
+class MockIceTransport : public IceTransportInterface {
public:
MockIceTransport(const std::string& name, int component)
: internal_(std::make_unique<cricket::FakeIceTransport>(
@@ -1407,7 +1400,7 @@
worker_thread_->SetName("PCWorkerThread", this);
RTC_CHECK(network_thread_->Start());
RTC_CHECK(worker_thread_->Start());
- webrtc::metrics::Reset();
+ metrics::Reset();
}
~PeerConnectionIntegrationBaseTest() {
@@ -1444,13 +1437,13 @@
// are connected. This is an important distinction. Once we have separate
// ICE and DTLS state, this check needs to use the DTLS state.
return (callee()->ice_connection_state() ==
- webrtc::PeerConnectionInterface::kIceConnectionConnected ||
+ PeerConnectionInterface::kIceConnectionConnected ||
callee()->ice_connection_state() ==
- webrtc::PeerConnectionInterface::kIceConnectionCompleted) &&
+ PeerConnectionInterface::kIceConnectionCompleted) &&
(caller()->ice_connection_state() ==
- webrtc::PeerConnectionInterface::kIceConnectionConnected ||
+ PeerConnectionInterface::kIceConnectionConnected ||
caller()->ice_connection_state() ==
- webrtc::PeerConnectionInterface::kIceConnectionCompleted);
+ PeerConnectionInterface::kIceConnectionCompleted);
}
// When `event_log_factory` is null, the default implementation of the event
@@ -1459,8 +1452,8 @@
const std::string& debug_name,
const PeerConnectionFactory::Options* options,
const RTCConfiguration* config,
- webrtc::PeerConnectionDependencies dependencies,
- std::unique_ptr<webrtc::FakeRtcEventLogFactory> event_log_factory,
+ PeerConnectionDependencies dependencies,
+ std::unique_ptr<FakeRtcEventLogFactory> event_log_factory,
bool reset_encoder_factory,
bool reset_decoder_factory,
bool create_media_engine = true) {
@@ -1490,10 +1483,10 @@
const std::string& debug_name,
const PeerConnectionFactory::Options* options,
const RTCConfiguration* config,
- webrtc::PeerConnectionDependencies dependencies) {
+ PeerConnectionDependencies dependencies) {
return CreatePeerConnectionWrapper(
debug_name, options, config, std::move(dependencies),
- std::make_unique<webrtc::FakeRtcEventLogFactory>(),
+ std::make_unique<FakeRtcEventLogFactory>(),
/*reset_encoder_factory=*/false,
/*reset_decoder_factory=*/false);
}
@@ -1514,17 +1507,17 @@
// callee PeerConnections.
SdpSemantics original_semantics = sdp_semantics_;
sdp_semantics_ = caller_semantics;
- caller_ = CreatePeerConnectionWrapper(
- "Caller", nullptr, nullptr, webrtc::PeerConnectionDependencies(nullptr),
- nullptr,
- /*reset_encoder_factory=*/false,
- /*reset_decoder_factory=*/false);
+ caller_ = CreatePeerConnectionWrapper("Caller", nullptr, nullptr,
+ PeerConnectionDependencies(nullptr),
+ nullptr,
+ /*reset_encoder_factory=*/false,
+ /*reset_decoder_factory=*/false);
sdp_semantics_ = callee_semantics;
- callee_ = CreatePeerConnectionWrapper(
- "Callee", nullptr, nullptr, webrtc::PeerConnectionDependencies(nullptr),
- nullptr,
- /*reset_encoder_factory=*/false,
- /*reset_decoder_factory=*/false);
+ callee_ = CreatePeerConnectionWrapper("Callee", nullptr, nullptr,
+ PeerConnectionDependencies(nullptr),
+ nullptr,
+ /*reset_encoder_factory=*/false,
+ /*reset_decoder_factory=*/false);
sdp_semantics_ = original_semantics;
return caller_ && callee_;
}
@@ -1532,24 +1525,24 @@
bool CreatePeerConnectionWrappersWithConfig(
const PeerConnectionInterface::RTCConfiguration& caller_config,
const PeerConnectionInterface::RTCConfiguration& callee_config) {
- caller_ = CreatePeerConnectionWrapper(
- "Caller", nullptr, &caller_config,
- webrtc::PeerConnectionDependencies(nullptr), nullptr,
- /*reset_encoder_factory=*/false,
- /*reset_decoder_factory=*/false);
- callee_ = CreatePeerConnectionWrapper(
- "Callee", nullptr, &callee_config,
- webrtc::PeerConnectionDependencies(nullptr), nullptr,
- /*reset_encoder_factory=*/false,
- /*reset_decoder_factory=*/false);
+ caller_ = CreatePeerConnectionWrapper("Caller", nullptr, &caller_config,
+ PeerConnectionDependencies(nullptr),
+ nullptr,
+ /*reset_encoder_factory=*/false,
+ /*reset_decoder_factory=*/false);
+ callee_ = CreatePeerConnectionWrapper("Callee", nullptr, &callee_config,
+ PeerConnectionDependencies(nullptr),
+ nullptr,
+ /*reset_encoder_factory=*/false,
+ /*reset_decoder_factory=*/false);
return caller_ && callee_;
}
bool CreatePeerConnectionWrappersWithConfigAndDeps(
const PeerConnectionInterface::RTCConfiguration& caller_config,
- webrtc::PeerConnectionDependencies caller_dependencies,
+ PeerConnectionDependencies caller_dependencies,
const PeerConnectionInterface::RTCConfiguration& callee_config,
- webrtc::PeerConnectionDependencies callee_dependencies) {
+ PeerConnectionDependencies callee_dependencies) {
caller_ =
CreatePeerConnectionWrapper("Caller", nullptr, &caller_config,
std::move(caller_dependencies), nullptr,
@@ -1566,16 +1559,16 @@
bool CreatePeerConnectionWrappersWithOptions(
const PeerConnectionFactory::Options& caller_options,
const PeerConnectionFactory::Options& callee_options) {
- caller_ = CreatePeerConnectionWrapper(
- "Caller", &caller_options, nullptr,
- webrtc::PeerConnectionDependencies(nullptr), nullptr,
- /*reset_encoder_factory=*/false,
- /*reset_decoder_factory=*/false);
- callee_ = CreatePeerConnectionWrapper(
- "Callee", &callee_options, nullptr,
- webrtc::PeerConnectionDependencies(nullptr), nullptr,
- /*reset_encoder_factory=*/false,
- /*reset_decoder_factory=*/false);
+ caller_ = CreatePeerConnectionWrapper("Caller", &caller_options, nullptr,
+ PeerConnectionDependencies(nullptr),
+ nullptr,
+ /*reset_encoder_factory=*/false,
+ /*reset_decoder_factory=*/false);
+ callee_ = CreatePeerConnectionWrapper("Callee", &callee_options, nullptr,
+ PeerConnectionDependencies(nullptr),
+ nullptr,
+ /*reset_encoder_factory=*/false,
+ /*reset_decoder_factory=*/false);
return caller_ && callee_;
}
@@ -1583,10 +1576,10 @@
PeerConnectionInterface::RTCConfiguration default_config;
caller_ = CreatePeerConnectionWrapperWithFakeRtcEventLog(
"Caller", nullptr, &default_config,
- webrtc::PeerConnectionDependencies(nullptr));
+ PeerConnectionDependencies(nullptr));
callee_ = CreatePeerConnectionWrapperWithFakeRtcEventLog(
"Callee", nullptr, &default_config,
- webrtc::PeerConnectionDependencies(nullptr));
+ PeerConnectionDependencies(nullptr));
return caller_ && callee_;
}
@@ -1596,7 +1589,7 @@
new FakeRTCCertificateGenerator());
cert_generator->use_alternate_key();
- webrtc::PeerConnectionDependencies dependencies(nullptr);
+ PeerConnectionDependencies dependencies(nullptr);
dependencies.cert_generator = std::move(cert_generator);
return CreatePeerConnectionWrapper("New Peer", nullptr, nullptr,
std::move(dependencies), nullptr,
@@ -1606,12 +1599,12 @@
bool CreateOneDirectionalPeerConnectionWrappers(bool caller_to_callee) {
caller_ = CreatePeerConnectionWrapper(
- "Caller", nullptr, nullptr, webrtc::PeerConnectionDependencies(nullptr),
+ "Caller", nullptr, nullptr, PeerConnectionDependencies(nullptr),
nullptr,
/*reset_encoder_factory=*/!caller_to_callee,
/*reset_decoder_factory=*/caller_to_callee);
callee_ = CreatePeerConnectionWrapper(
- "Callee", nullptr, nullptr, webrtc::PeerConnectionDependencies(nullptr),
+ "Callee", nullptr, nullptr, PeerConnectionDependencies(nullptr),
nullptr,
/*reset_encoder_factory=*/caller_to_callee,
/*reset_decoder_factory=*/!caller_to_callee);
@@ -1619,18 +1612,18 @@
}
bool CreatePeerConnectionWrappersWithoutMediaEngine() {
- caller_ = CreatePeerConnectionWrapper(
- "Caller", nullptr, nullptr, webrtc::PeerConnectionDependencies(nullptr),
- nullptr,
- /*reset_encoder_factory=*/false,
- /*reset_decoder_factory=*/false,
- /*create_media_engine=*/false);
- callee_ = CreatePeerConnectionWrapper(
- "Callee", nullptr, nullptr, webrtc::PeerConnectionDependencies(nullptr),
- nullptr,
- /*reset_encoder_factory=*/false,
- /*reset_decoder_factory=*/false,
- /*create_media_engine=*/false);
+ caller_ = CreatePeerConnectionWrapper("Caller", nullptr, nullptr,
+ PeerConnectionDependencies(nullptr),
+ nullptr,
+ /*reset_encoder_factory=*/false,
+ /*reset_decoder_factory=*/false,
+ /*create_media_engine=*/false);
+ callee_ = CreatePeerConnectionWrapper("Callee", nullptr, nullptr,
+ PeerConnectionDependencies(nullptr),
+ nullptr,
+ /*reset_encoder_factory=*/false,
+ /*reset_decoder_factory=*/false,
+ /*create_media_engine=*/false);
return caller_ && callee_;
}
@@ -1700,7 +1693,7 @@
// Messages may get lost on the unreliable DataChannel, so we send multiple
// times to avoid test flakiness.
- void SendRtpDataWithRetries(webrtc::DataChannelInterface* dc,
+ void SendRtpDataWithRetries(DataChannelInterface* dc,
const std::string& data,
int retries) {
for (int i = 0; i < retries; ++i) {
diff --git a/pc/test/mock_peer_connection_observers.h b/pc/test/mock_peer_connection_observers.h
index e9d97a9..6222ef7 100644
--- a/pc/test/mock_peer_connection_observers.h
+++ b/pc/test/mock_peer_connection_observers.h
@@ -254,7 +254,7 @@
};
class MockCreateSessionDescriptionObserver
- : public webrtc::CreateSessionDescriptionObserver {
+ : public CreateSessionDescriptionObserver {
public:
MockCreateSessionDescriptionObserver()
: called_(false),
@@ -266,7 +266,7 @@
error_ = "";
desc_.reset(desc);
}
- void OnFailure(webrtc::RTCError error) override {
+ void OnFailure(RTCError error) override {
MutexLock lock(&mutex_);
called_ = true;
error_ = error.message();
@@ -295,8 +295,7 @@
std::unique_ptr<SessionDescriptionInterface> desc_ RTC_GUARDED_BY(mutex_);
};
-class MockSetSessionDescriptionObserver
- : public webrtc::SetSessionDescriptionObserver {
+class MockSetSessionDescriptionObserver : public SetSessionDescriptionObserver {
public:
static rtc::scoped_refptr<MockSetSessionDescriptionObserver> Create() {
return rtc::make_ref_counted<MockSetSessionDescriptionObserver>();
@@ -312,7 +311,7 @@
called_ = true;
error_ = "";
}
- void OnFailure(webrtc::RTCError error) override {
+ void OnFailure(RTCError error) override {
MutexLock lock(&mutex_);
called_ = true;
error_ = error.message();
@@ -375,14 +374,14 @@
absl::optional<RTCError> error_;
};
-class MockDataChannelObserver : public webrtc::DataChannelObserver {
+class MockDataChannelObserver : public DataChannelObserver {
public:
struct Message {
std::string data;
bool binary;
};
- explicit MockDataChannelObserver(webrtc::DataChannelInterface* channel)
+ explicit MockDataChannelObserver(DataChannelInterface* channel)
: channel_(channel) {
channel_->RegisterObserver(this);
states_.push_back(channel_->state());
@@ -419,12 +418,12 @@
}
private:
- rtc::scoped_refptr<webrtc::DataChannelInterface> channel_;
+ rtc::scoped_refptr<DataChannelInterface> channel_;
std::vector<DataChannelInterface::DataState> states_;
std::vector<Message> messages_;
};
-class MockStatsObserver : public webrtc::StatsObserver {
+class MockStatsObserver : public StatsObserver {
public:
MockStatsObserver() : called_(false), stats_() {}
virtual ~MockStatsObserver() {}
@@ -576,7 +575,7 @@
};
// Helper class that just stores the report from the callback.
-class MockRTCStatsCollectorCallback : public webrtc::RTCStatsCollectorCallback {
+class MockRTCStatsCollectorCallback : public RTCStatsCollectorCallback {
public:
rtc::scoped_refptr<const RTCStatsReport> report() { return report_; }
diff --git a/pc/test/rtp_transport_test_util.h b/pc/test/rtp_transport_test_util.h
index 8aeaf07..563014f 100644
--- a/pc/test/rtp_transport_test_util.h
+++ b/pc/test/rtp_transport_test_util.h
@@ -33,9 +33,7 @@
rtp_transport->SubscribeReadyToSend(
this, [this](bool arg) { OnReadyToSend(arg); });
rtp_transport->SetUnDemuxableRtpPacketReceivedHandler(
- [this](webrtc::RtpPacketReceived& packet) {
- OnUndemuxableRtpPacket(packet);
- });
+ [this](RtpPacketReceived& packet) { OnUndemuxableRtpPacket(packet); });
rtp_transport->SubscribeSentPacket(this,
[this](const rtc::SentPacket& packet) {
sent_packet_count_++;
diff --git a/pc/test/svc_e2e_tests.cc b/pc/test/svc_e2e_tests.cc
index ae35c7f..412027b 100644
--- a/pc/test/svc_e2e_tests.cc
+++ b/pc/test/svc_e2e_tests.cc
@@ -160,10 +160,9 @@
// encoder and decoder level.
class SvcVideoQualityAnalyzer : public DefaultVideoQualityAnalyzer {
public:
- using SpatialTemporalLayerCounts =
- webrtc::flat_map<int, webrtc::flat_map<int, int>>;
+ using SpatialTemporalLayerCounts = flat_map<int, flat_map<int, int>>;
- explicit SvcVideoQualityAnalyzer(webrtc::Clock* clock)
+ explicit SvcVideoQualityAnalyzer(Clock* clock)
: DefaultVideoQualityAnalyzer(clock,
test::GetGlobalMetricsLogger(),
DefaultVideoQualityAnalyzerOptions{
@@ -315,9 +314,9 @@
if (UseDependencyDescriptor()) {
trials += "WebRTC-DependencyDescriptorAdvertised/Enabled/";
}
- webrtc::test::ScopedFieldTrials override_trials(AppendFieldTrials(trials));
+ test::ScopedFieldTrials override_trials(AppendFieldTrials(trials));
std::unique_ptr<NetworkEmulationManager> network_emulation_manager =
- CreateNetworkEmulationManager(webrtc::TimeMode::kSimulated);
+ CreateNetworkEmulationManager(TimeMode::kSimulated);
auto analyzer = std::make_unique<SvcVideoQualityAnalyzer>(
network_emulation_manager->time_controller()->GetClock());
SvcVideoQualityAnalyzer* analyzer_ptr = analyzer.get();
diff --git a/pc/video_rtp_receiver_unittest.cc b/pc/video_rtp_receiver_unittest.cc
index 5ff7360..e972917 100644
--- a/pc/video_rtp_receiver_unittest.cc
+++ b/pc/video_rtp_receiver_unittest.cc
@@ -94,7 +94,7 @@
[&]() { receiver_->SetMediaChannel(media_channel); });
}
- webrtc::VideoTrackSourceInterface* Source() {
+ VideoTrackSourceInterface* Source() {
return receiver_->streams()[0]->FindVideoTrack("receiver")->GetSource();
}
diff --git a/pc/video_rtp_track_source_unittest.cc b/pc/video_rtp_track_source_unittest.cc
index 13728c7..55632ce 100644
--- a/pc/video_rtp_track_source_unittest.cc
+++ b/pc/video_rtp_track_source_unittest.cc
@@ -109,11 +109,11 @@
class TestFrame : public RecordableEncodedFrame {
public:
- rtc::scoped_refptr<const webrtc::EncodedImageBufferInterface> encoded_buffer()
+ rtc::scoped_refptr<const EncodedImageBufferInterface> encoded_buffer()
const override {
return nullptr;
}
- absl::optional<webrtc::ColorSpace> color_space() const override {
+ absl::optional<ColorSpace> color_space() const override {
return absl::nullopt;
}
VideoCodecType codec() const override { return kVideoCodecGeneric; }
diff --git a/pc/video_track.h b/pc/video_track.h
index 13a51c4..e504182 100644
--- a/pc/video_track.h
+++ b/pc/video_track.h
@@ -70,7 +70,7 @@
// Implements ObserverInterface. Observes `video_source_` state.
void OnChanged() override;
- RTC_NO_UNIQUE_ADDRESS webrtc::SequenceChecker signaling_thread_;
+ RTC_NO_UNIQUE_ADDRESS SequenceChecker signaling_thread_;
rtc::Thread* const worker_thread_;
const rtc::scoped_refptr<
VideoTrackSourceProxyWithInternal<VideoTrackSourceInterface>>
diff --git a/pc/video_track_source_proxy.h b/pc/video_track_source_proxy.h
index 8500a98..40d2423 100644
--- a/pc/video_track_source_proxy.h
+++ b/pc/video_track_source_proxy.h
@@ -52,7 +52,7 @@
rtc::VideoSinkInterface<RecordableEncodedFrame>*)
PROXY_SECONDARY_METHOD1(void,
ProcessConstraints,
- const webrtc::VideoTrackSourceConstraints&)
+ const VideoTrackSourceConstraints&)
END_PROXY_MAP(VideoTrackSource)
} // namespace webrtc
diff --git a/pc/webrtc_sdp.cc b/pc/webrtc_sdp.cc
index 7d896e2..da024ea 100644
--- a/pc/webrtc_sdp.cc
+++ b/pc/webrtc_sdp.cc
@@ -2626,7 +2626,7 @@
int* msid_signaling,
TransportDescription* transport,
std::vector<std::unique_ptr<JsepIceCandidate>>* candidates,
- webrtc::SdpParseError* error) {
+ SdpParseError* error) {
std::unique_ptr<MediaContentDescription> media_desc;
if (media_type == cricket::MediaType::MEDIA_TYPE_AUDIO) {
media_desc = std::make_unique<AudioContentDescription>();
diff --git a/tools_webrtc/remove_extra_namespace.py b/tools_webrtc/remove_extra_namespace.py
new file mode 100755
index 0000000..21ac2d1
--- /dev/null
+++ b/tools_webrtc/remove_extra_namespace.py
@@ -0,0 +1,93 @@
+#!/usr/bin/env vpython3
+
+# Copyright (c) 2023 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+"""Remove extra namespace qualifications
+
+Looks for names that don't need to be qualified by namespace, and deletes
+the qualifier.
+
+Depends on namespace names being properly formatted
+"""
+import os
+import glob
+import sys
+import re
+import argparse
+
+
+def remove_extra_namespace_from_file(namespace, filename):
+ print('Processing namespace', namespace, 'file', filename)
+ with open(filename) as file:
+ newfile = open(filename + '.NEW', 'w')
+ namespaces = []
+ changes = 0
+ for line in file:
+ match = re.match(r'namespace (\S+) {', line)
+ if match is not None:
+ namespaces.insert(0, match.group(1))
+ newfile.write(line)
+ continue
+ match = re.match(r'}\s+// namespace (\S+)$', line)
+ if match is not None:
+ if match.group(1) != namespaces[0]:
+ print('Namespace mismatch')
+ raise RuntimeError('Namespace mismatch')
+ del namespaces[0]
+ newfile.write(line)
+ continue
+ # Remove namespace usage. Only replacing when target
+ # namespace is the innermost namespace.
+ if len(namespaces) > 0 and namespaces[0] == namespace:
+ # Note that in namespace foo, we match neither ::foo::name
+ # nor morefoo::name
+ # Neither do we match foo:: when it is not followed by
+ # an identifier character.
+ usage_re = r'(?<=[^a-z:]){}::(?=[a-zA-Z])'.format(
+ namespaces[0])
+ if re.search(usage_re, line):
+ line = re.sub(usage_re, '', line)
+ changes += 1
+ newfile.write(line)
+ if changes > 0:
+ print('Made', changes, 'changes to', filename)
+ os.remove(filename)
+ os.rename(filename + '.NEW', filename)
+ else:
+ os.remove(filename + '.NEW')
+
+
+def remove_extra_namespace_from_files(namespace, files):
+ for file in files:
+ if os.path.isfile(file):
+ if re.search(r'\.(h|cc)$', file):
+ remove_extra_namespace_from_file(namespace, file)
+ elif os.path.isdir(file):
+ if file in ('third_party', 'out'):
+ continue
+ subfiles = glob.glob(file + '/*')
+ remove_extra_namespace_from_files(namespace, subfiles)
+ else:
+ print(file, 'is not a file or directory, ignoring')
+
+
+def main():
+ parser = argparse.ArgumentParser(
+ prog='remove_extra_namespace.py',
+ description=__doc__.strip().splitlines()[0],
+ epilog=''.join(__doc__.splitlines(True)[1:]),
+ formatter_class=argparse.RawDescriptionHelpFormatter,
+ )
+ parser.add_argument('--namespace')
+ parser.add_argument('files', nargs=argparse.REMAINDER)
+ args = parser.parse_args()
+ return remove_extra_namespace_from_files(args.namespace, args.files)
+
+
+if __name__ == '__main__':
+ sys.exit(main())
