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}

pc/peer_connection.cc

@@ -1481,26 +1481,6 @@ void PeerConnection::SetAudioRecording(bool recording) {
   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()) {

pc/peer_connection.h

@@ -154,20 +154,6 @@ class PeerConnection : public PeerConnectionInternal,
       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;

pc/peer_connection_integrationtest.cc

@@ -1950,76 +1950,6 @@ TEST_P(PeerConnectionIntegrationTest,
   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,

pc/rtp_transmission_manager.cc

@@ -440,17 +440,6 @@ void RtpTransmissionManager::RemoveVideoTrack(VideoTrackInterface* track,
   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) {

pc/rtp_transmission_manager.h

@@ -136,10 +136,6 @@ class RtpTransmissionManager : public RtpSenderBase::SetStreamsObserver,
   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.