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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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25
webrtc/voice_engine/channel.cc
View File

@ -868,6 +868,7 @@ Channel::Channel(int32_t channelId,
_callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
_instanceId(instanceId),
_channelId(channelId),
rtp_header_parser_(RtpHeaderParser::Create()),
_audioCodingModule(*AudioCodingModule::Create(
VoEModuleId(instanceId, channelId))),
_rtpDumpIn(*RtpDump::CreateRtpDump()),
@ -2128,12 +2129,20 @@ int32_t Channel::ReceivedRTPPacket(const int8_t* data, int32_t length) {
VoEId(_instanceId,_channelId),
"Channel::SendPacket() RTP dump to input file failed");
}
RTPHeader header;
if (!rtp_header_parser_->Parse(reinterpret_cast<const uint8_t*>(data),
static_cast<uint16_t>(length), &header)) {
WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideo,
VoEId(_instanceId,_channelId),
"IncomingPacket invalid RTP header");
return -1;
}
// Deliver RTP packet to RTP/RTCP module for parsing
// The packet will be pushed back to the channel thru the
// OnReceivedPayloadData callback so we don't push it to the ACM here
if (_rtpRtcpModule->IncomingPacket((const uint8_t*)data,
(uint16_t)length) == -1) {
if (_rtpRtcpModule->IncomingRtpPacket(reinterpret_cast<const uint8_t*>(data),
static_cast<uint16_t>(length),
header) == -1) {
_engineStatisticsPtr->SetLastError(
VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceWarning,
"Channel::IncomingRTPPacket() RTP packet is invalid");
@ -2156,8 +2165,8 @@ int32_t Channel::ReceivedRTCPPacket(const int8_t* data, int32_t length) {
}
// Deliver RTCP packet to RTP/RTCP module for parsing
if (_rtpRtcpModule->IncomingPacket((const uint8_t*)data,
(uint16_t)length) == -1) {
if (_rtpRtcpModule->IncomingRtcpPacket((const uint8_t*)data,
(uint16_t)length) == -1) {
_engineStatisticsPtr->SetLastError(
VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceWarning,
"Channel::IncomingRTPPacket() RTCP packet is invalid");
@ -3699,6 +3708,12 @@ Channel::SetRTPAudioLevelIndicationStatus(bool enable, unsigned char ID)
}
_includeAudioLevelIndication = enable;
if (enable) {
rtp_header_parser_->RegisterRtpHeaderExtension(kRtpExtensionAudioLevel,
ID);
} else {
rtp_header_parser_->DeregisterRtpHeaderExtension(kRtpExtensionAudioLevel);
}
return _rtpRtcpModule->SetRTPAudioLevelIndicationStatus(enable, ID);
}
int