Index: webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc |
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc b/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc |
index 092ac1cc4989137d520fa7938728819ee343630c..5aa7e27d2391e75ede72cba6c74bf318937121bc 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc |
@@ -389,15 +389,6 @@ void RtpPacketizerH264::NextFragmentPacket(RtpPacketToSend* rtp_packet) { |
packets_.pop(); |
} |
-ProtectionType RtpPacketizerH264::GetProtectionType() { |
- return kProtectedPacket; |
-} |
- |
-StorageType RtpPacketizerH264::GetStorageType( |
- uint32_t retransmission_settings) { |
- return kAllowRetransmission; |
-} |
- |
std::string RtpPacketizerH264::ToString() { |
return "RtpPacketizerH264"; |
} |