Bug 1828517 (MOZ) - handle upstream removing AudioReceiveStreamInterface::GetRtpExtensions

media/webrtc/signaling/gtest/MockCall.cpp

@@ -29,12 +29,6 @@ void MockAudioReceiveStream::SetRtpExtensions(
       std::move(extensions);
 }
 
-const std::vector<webrtc::RtpExtension>&
-MockAudioReceiveStream::GetRtpExtensions() const {
-  static std::vector<webrtc::RtpExtension> rtpExtensions;
-  return rtpExtensions;
-}
-
 webrtc::RtpHeaderExtensionMap MockAudioReceiveStream::GetRtpExtensionMap()
     const {
   return webrtc::RtpHeaderExtensionMap();

media/webrtc/signaling/gtest/MockCall.h

@@ -94,7 +94,6 @@ class MockAudioReceiveStream : public webrtc::AudioReceiveStreamInterface {
   void SetFrameDecryptor(rtc::scoped_refptr<webrtc::FrameDecryptorInterface>
                              frame_decryptor) override {}
   void SetRtpExtensions(std::vector<webrtc::RtpExtension> extensions) override;
-  const std::vector<webrtc::RtpExtension>& GetRtpExtensions() const override;
   webrtc::RtpHeaderExtensionMap GetRtpExtensionMap() const override;
   bool SetBaseMinimumPlayoutDelayMs(int delay_ms) override { return false; }
   int GetBaseMinimumPlayoutDelayMs() const override { return 0; }