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Remove obsolete GetRemoteAudioSSL* functions.

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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}
This commit is contained in:
Harald Alvestrand
2020-10-30 06:44:55 +00:00
committed by Commit Bot
parent 9f52b9dc4e
commit 4efa9d0a5f
5 changed files with 0 additions and 119 deletions
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20
pc/peer_connection.cc
View File

@ -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()) {

14
pc/peer_connection.h
View File

@ -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;

70
pc/peer_connection_integrationtest.cc
View File

@ -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,

11
pc/rtp_transmission_manager.cc
View File

@ -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) {

4
pc/rtp_transmission_manager.h
View File

@ -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.