| 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,
|
|
|