Breaking out RTP header parsing from the RTP module.
This is the first step in order to move bandwidth estimation closer to the network. The goal is to have RTP header parsing and bandwidth estimation before voice and video engine, and have a joint estimate for audio and video.
Moving bandwidth estimation before the RTP module is also required for RTX.
TEST=vie_auto_test, voe_auto_test, trybots.
BUG=1811
R=andresp@webrtc.org, henrika@webrtc.org, mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1545004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4129 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h b/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
index 0a36a9e..0313076 100644
--- a/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
+++ b/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
@@ -57,10 +57,6 @@
int32_t(const VideoCodec& videoCodec, int8_t* plType));
MOCK_METHOD1(DeRegisterReceivePayload,
int32_t(const int8_t payloadType));
- MOCK_METHOD2(RegisterReceiveRtpHeaderExtension,
- int32_t(const RTPExtensionType type, const uint8_t id));
- MOCK_METHOD1(DeregisterReceiveRtpHeaderExtension,
- int32_t(const RTPExtensionType type));
MOCK_CONST_METHOD0(RemoteTimestamp,
uint32_t());
MOCK_CONST_METHOD0(LocalTimeOfRemoteTimeStamp,
@@ -81,8 +77,11 @@
int32_t(bool* enable, uint32_t* ssrc, int* payload_type));
MOCK_METHOD1(SetRtxReceivePayloadType,
void(int));
- MOCK_METHOD2(IncomingPacket,
- int32_t(const uint8_t* incomingPacket, const uint16_t packetLength));
+ MOCK_METHOD3(IncomingRtpPacket,
+ int32_t(const uint8_t* incomingPacket, const uint16_t packetLength,
+ const webrtc::RTPHeader& header));
+ MOCK_METHOD2(IncomingRtcpPacket,
+ int32_t(const uint8_t* incomingPacket, uint16_t packetLength));
MOCK_METHOD4(IncomingAudioNTP,
int32_t(const uint32_t audioReceivedNTPsecs,
const uint32_t audioReceivedNTPfrac,