Remove obsolete GetRemoteAudioSSL* functions.
Bug: webrtc:12054
Change-Id: I56d198cfa2c336155c5173ccd524107d12e6a382
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190921
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32528}
diff --git a/pc/peer_connection.cc b/pc/peer_connection.cc
index 4a6c85c..8faac3b 100644
--- a/pc/peer_connection.cc
+++ b/pc/peer_connection.cc
@@ -1481,26 +1481,6 @@
audio_state->SetRecording(recording);
}
-std::unique_ptr<rtc::SSLCertificate>
-PeerConnection::GetRemoteAudioSSLCertificate() {
- std::unique_ptr<rtc::SSLCertChain> chain = GetRemoteAudioSSLCertChain();
- if (!chain || !chain->GetSize()) {
- return nullptr;
- }
- return chain->Get(0).Clone();
-}
-
-std::unique_ptr<rtc::SSLCertChain>
-PeerConnection::GetRemoteAudioSSLCertChain() {
- RTC_DCHECK_RUN_ON(signaling_thread());
- auto audio_transceiver = rtp_manager()->GetFirstAudioTransceiver();
- if (!audio_transceiver || !audio_transceiver->internal()->channel()) {
- return nullptr;
- }
- return transport_controller_->GetRemoteSSLCertChain(
- audio_transceiver->internal()->channel()->transport_name());
-}
-
void PeerConnection::AddAdaptationResource(
rtc::scoped_refptr<Resource> resource) {
if (!worker_thread()->IsCurrent()) {
diff --git a/pc/peer_connection.h b/pc/peer_connection.h
index 35567c1..d928b3e 100644
--- a/pc/peer_connection.h
+++ b/pc/peer_connection.h
@@ -154,20 +154,6 @@
cricket::MediaType media_type,
const RtpTransceiverInit& init) override;
- // Gets the DTLS SSL certificate associated with the audio transport on the
- // remote side. This will become populated once the DTLS connection with the
- // peer has been completed, as indicated by the ICE connection state
- // transitioning to kIceConnectionCompleted.
- // Deprecated - users should insted query the DTLS transpport for this info.
- // See https://www.w3.org/TR/webrtc/#rtcdtlstransport-interface
-
- RTC_DEPRECATED std::unique_ptr<rtc::SSLCertificate>
- GetRemoteAudioSSLCertificate();
-
- // Version of the above method that returns the full certificate chain.
- RTC_DEPRECATED std::unique_ptr<rtc::SSLCertChain>
- GetRemoteAudioSSLCertChain();
-
rtc::scoped_refptr<RtpSenderInterface> CreateSender(
const std::string& kind,
const std::string& stream_id) override;
diff --git a/pc/peer_connection_integrationtest.cc b/pc/peer_connection_integrationtest.cc
index a72dbc6..53e0f6d 100644
--- a/pc/peer_connection_integrationtest.cc
+++ b/pc/peer_connection_integrationtest.cc
@@ -1950,76 +1950,6 @@
ASSERT_TRUE(ExpectNewFrames(media_expectations));
}
-// Tests that the GetRemoteAudioSSLCertificate method returns the remote DTLS
-// certificate once the DTLS handshake has finished.
-TEST_P(PeerConnectionIntegrationTest,
- GetRemoteAudioSSLCertificateReturnsExchangedCertificate) {
- auto GetRemoteAudioSSLCertificate = [](PeerConnectionWrapper* wrapper) {
- auto pci = reinterpret_cast<PeerConnectionProxy*>(wrapper->pc());
- auto pc = reinterpret_cast<PeerConnection*>(pci->internal());
- return pc->GetRemoteAudioSSLCertificate();
- };
- auto GetRemoteAudioSSLCertChain = [](PeerConnectionWrapper* wrapper) {
- auto pci = reinterpret_cast<PeerConnectionProxy*>(wrapper->pc());
- auto pc = reinterpret_cast<PeerConnection*>(pci->internal());
- return pc->GetRemoteAudioSSLCertChain();
- };
-
- auto caller_cert = rtc::RTCCertificate::FromPEM(kRsaPems[0]);
- auto callee_cert = rtc::RTCCertificate::FromPEM(kRsaPems[1]);
-
- // Configure each side with a known certificate so they can be compared later.
- PeerConnectionInterface::RTCConfiguration caller_config;
- caller_config.enable_dtls_srtp.emplace(true);
- caller_config.certificates.push_back(caller_cert);
- PeerConnectionInterface::RTCConfiguration callee_config;
- callee_config.enable_dtls_srtp.emplace(true);
- callee_config.certificates.push_back(callee_cert);
- ASSERT_TRUE(
- CreatePeerConnectionWrappersWithConfig(caller_config, callee_config));
- ConnectFakeSignaling();
-
- // When first initialized, there should not be a remote SSL certificate (and
- // calling this method should not crash).
- EXPECT_EQ(nullptr, GetRemoteAudioSSLCertificate(caller()));
- EXPECT_EQ(nullptr, GetRemoteAudioSSLCertificate(callee()));
- EXPECT_EQ(nullptr, GetRemoteAudioSSLCertChain(caller()));
- EXPECT_EQ(nullptr, GetRemoteAudioSSLCertChain(callee()));
-
- caller()->AddAudioTrack();
- callee()->AddAudioTrack();
- caller()->CreateAndSetAndSignalOffer();
- ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
- ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
-
- // Once DTLS has been connected, each side should return the other's SSL
- // certificate when calling GetRemoteAudioSSLCertificate.
-
- auto caller_remote_cert = GetRemoteAudioSSLCertificate(caller());
- ASSERT_TRUE(caller_remote_cert);
- EXPECT_EQ(callee_cert->GetSSLCertificate().ToPEMString(),
- caller_remote_cert->ToPEMString());
-
- auto callee_remote_cert = GetRemoteAudioSSLCertificate(callee());
- ASSERT_TRUE(callee_remote_cert);
- EXPECT_EQ(caller_cert->GetSSLCertificate().ToPEMString(),
- callee_remote_cert->ToPEMString());
-
- auto caller_remote_cert_chain = GetRemoteAudioSSLCertChain(caller());
- ASSERT_TRUE(caller_remote_cert_chain);
- ASSERT_EQ(1U, caller_remote_cert_chain->GetSize());
- auto remote_cert = &caller_remote_cert_chain->Get(0);
- EXPECT_EQ(callee_cert->GetSSLCertificate().ToPEMString(),
- remote_cert->ToPEMString());
-
- auto callee_remote_cert_chain = GetRemoteAudioSSLCertChain(callee());
- ASSERT_TRUE(callee_remote_cert_chain);
- ASSERT_EQ(1U, callee_remote_cert_chain->GetSize());
- remote_cert = &callee_remote_cert_chain->Get(0);
- EXPECT_EQ(caller_cert->GetSSLCertificate().ToPEMString(),
- remote_cert->ToPEMString());
-}
-
// This test sets up a call between two parties with a source resolution of
// 1280x720 and verifies that a 16:9 aspect ratio is received.
TEST_P(PeerConnectionIntegrationTest,
diff --git a/pc/rtp_transmission_manager.cc b/pc/rtp_transmission_manager.cc
index 5c0e4d9..a71ed53 100644
--- a/pc/rtp_transmission_manager.cc
+++ b/pc/rtp_transmission_manager.cc
@@ -440,17 +440,6 @@
GetVideoTransceiver()->internal()->RemoveSender(sender);
}
-rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
-RtpTransmissionManager::GetFirstAudioTransceiver() const {
- RTC_DCHECK_RUN_ON(signaling_thread());
- for (auto transceiver : transceivers_.List()) {
- if (transceiver->media_type() == cricket::MEDIA_TYPE_AUDIO) {
- return transceiver;
- }
- }
- return nullptr;
-}
-
void RtpTransmissionManager::CreateAudioReceiver(
MediaStreamInterface* stream,
const RtpSenderInfo& remote_sender_info) {
diff --git a/pc/rtp_transmission_manager.h b/pc/rtp_transmission_manager.h
index 7aa410c..de83fb2 100644
--- a/pc/rtp_transmission_manager.h
+++ b/pc/rtp_transmission_manager.h
@@ -136,10 +136,6 @@
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
GetVideoTransceiver() const;
- // Gets the first audio transceiver.
- rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
- GetFirstAudioTransceiver() const;
-
// Add an audio track, reusing or creating the sender.
void AddAudioTrack(AudioTrackInterface* track, MediaStreamInterface* stream);
// Plan B: Remove an audio track, removing the sender.
