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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
Download Patch File Download Diff File Expand all files Collapse all files

28
webrtc/modules/video_coding/main/test/rtp_player.cc
View File

@ -13,8 +13,8 @@
#include <cstdio>
#include <map>
#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "webrtc/modules/video_coding/main/source/internal_defines.h"
#include "webrtc/modules/video_coding/main/test/pcap_file_reader.h"
#include "webrtc/modules/video_coding/main/test/rtp_file_reader.h"
@ -229,7 +229,7 @@ class SsrcHandlers {
handler->rtp_module_->SetRTCPStatus(kRtcpNonCompound);
handler->rtp_module_->SetREMBStatus(true);
handler->rtp_module_->SetSSRCFilter(true, ssrc);
handler->rtp_module_->RegisterReceiveRtpHeaderExtension(
handler->rtp_header_parser_->RegisterRtpHeaderExtension(
kRtpExtensionTransmissionTimeOffset,
kDefaultTransmissionTimeOffsetExtensionId);
@ -257,7 +257,13 @@ class SsrcHandlers {
void IncomingPacket(const uint8_t* data, uint32_t length) {
for (HandlerMapIt it = handlers_.begin(); it != handlers_.end(); ++it) {
it->second->rtp_module_->IncomingPacket(data, length);
if (it->second->rtp_header_parser_->IsRtcp(data, length)) {
it->second->rtp_module_->IncomingRtcpPacket(data, length);
} else {
RTPHeader header;
it->second->rtp_header_parser_->Parse(data, length, &header);
it->second->rtp_module_->IncomingRtpPacket(data, length, header);
}
}
}
@ -266,7 +272,8 @@ class SsrcHandlers {
public:
Handler(uint32_t ssrc, const PayloadTypes& payload_types,
LostPackets* lost_packets)
: rtp_module_(),
: rtp_header_parser_(RtpHeaderParser::Create()),
rtp_module_(),
payload_sink_(),
ssrc_(ssrc),
payload_types_(payload_types),
@ -288,6 +295,7 @@ class SsrcHandlers {
return payload_types_;
}
scoped_ptr<RtpHeaderParser> rtp_header_parser_;
scoped_ptr<RtpRtcp> rtp_module_;
scoped_ptr<PayloadSinkInterface> payload_sink_;
@ -421,13 +429,13 @@ class RtpPlayerImpl : public RtpPlayerInterface {
assert(data);
assert(length > 0);
ModuleRTPUtility::RTPHeaderParser rtp_header_parser(data, length);
if (!rtp_header_parser.RTCP()) {
WebRtcRTPHeader header;
if (!rtp_header_parser.Parse(header, NULL)) {
scoped_ptr<RtpHeaderParser> rtp_header_parser(RtpHeaderParser::Create());
if (!rtp_header_parser->IsRtcp(data, length)) {
RTPHeader header;
if (!rtp_header_parser->Parse(data, length, &header)) {
return -1;
}
uint32_t ssrc = header.header.ssrc;
uint32_t ssrc = header.ssrc;
if (ssrc_handlers_.RegisterSsrc(ssrc, &lost_packets_) < 0) {
DEBUG_LOG1("Unable to register ssrc: %d", ssrc);
return -1;
@ -436,7 +444,7 @@ class RtpPlayerImpl : public RtpPlayerInterface {
if (no_loss_startup_ > 0) {
no_loss_startup_--;
} else if ((rand() + 1.0)/(RAND_MAX + 1.0) < loss_rate_) {
uint16_t seq_num = header.header.sequenceNumber;
uint16_t seq_num = header.sequenceNumber;
lost_packets_.AddPacket(new RawRtpPacket(data, length, ssrc, seq_num));
DEBUG_LOG1("Dropped packet: %d!", header.header.sequenceNumber);
return 0;