Index: webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h |
diff --git a/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h b/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h |
index 0e81e4dc6ef128f6951277996ba0ac8bb2cd1d5b..fc93655aa5553a7af8cf4dc1ced4ce66ae952dcf 100644 |
--- a/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h |
+++ b/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h |
@@ -53,12 +53,10 @@ class MockRtpRtcp : public RtpRtcp { |
uint32_t audio_rtcp_arrival_time_frac)); |
MOCK_METHOD0(InitSender, int32_t()); |
MOCK_METHOD1(RegisterSendTransport, int32_t(Transport* outgoing_transport)); |
- MOCK_METHOD1(SetMaxTransferUnit, int32_t(uint16_t size)); |
- MOCK_METHOD3(SetTransportOverhead, |
- int32_t(bool tcp, bool ipv6, uint8_t authentication_overhead)); |
+ MOCK_METHOD1(SetMaxRtpPacketSize, void(size_t size)); |
MOCK_METHOD1(SetTransportOverhead, void(int transport_overhead_per_packet)); |
- MOCK_CONST_METHOD0(MaxPayloadLength, uint16_t()); |
- MOCK_CONST_METHOD0(MaxDataPayloadLength, uint16_t()); |
+ MOCK_CONST_METHOD0(MaxPayloadSize, size_t()); |
+ MOCK_CONST_METHOD0(MaxRtpPacketSize, size_t()); |
MOCK_METHOD1(RegisterSendPayload, int32_t(const CodecInst& voice_codec)); |
MOCK_METHOD1(RegisterSendPayload, int32_t(const VideoCodec& video_codec)); |
MOCK_METHOD2(RegisterVideoSendPayload, |