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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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4
webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
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@ -79,9 +79,9 @@ int32_t RTPReceiverVideo::ParseRtpPacket(
"seqnum", rtp_header->header.sequenceNumber,
"timestamp", rtp_header->header.timestamp);
const uint8_t* payload_data =
ModuleRTPUtility::GetPayloadData(rtp_header, packet);
ModuleRTPUtility::GetPayloadData(rtp_header->header, packet);
const uint16_t payload_data_length =
ModuleRTPUtility::GetPayloadDataLength(rtp_header, packet_length);
ModuleRTPUtility::GetPayloadDataLength(rtp_header->header, packet_length);
return ParseVideoCodecSpecific(rtp_header,
payload_data,
payload_data_length,