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Breaking out RTP header parsing from the RTP module.

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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@4129 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
stefan@webrtc.org
2013-05-29 12:12:51 +00:00
parent 1ecee9a15a
commit a5cb98cbbd
37 changed files with 515 additions and 380 deletions
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19
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
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@ -70,14 +70,6 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
virtual int32_t DeRegisterReceivePayload(
const int8_t payload_type);
// Register RTP header extension.
virtual int32_t RegisterReceiveRtpHeaderExtension(
const RTPExtensionType type,
const uint8_t id);
virtual int32_t DeregisterReceiveRtpHeaderExtension(
const RTPExtensionType type);
// Get the currently configured SSRC filter.
virtual int32_t SSRCFilter(uint32_t& allowed_ssrc) const;
@ -108,9 +100,14 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
virtual void SetRtxReceivePayloadType(int payload_type);
// Called by the network module when we receive a packet.
virtual int32_t IncomingPacket(const uint8_t* incoming_packet,
const uint16_t packet_length);
// Called when we receive an RTP packet.
virtual int32_t IncomingRtpPacket(const uint8_t* incoming_packet,
const uint16_t packet_length,
const RTPHeader& parsed_rtp_header);
// Called when we receive an RTCP packet.
virtual int32_t IncomingRtcpPacket(const uint8_t* incoming_packet,
uint16_t incoming_packet_length);
// Sender part.