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Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream.

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BUG=
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5293 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
solenberg@webrtc.org
2013-12-13 23:57:54 +00:00
parent e1bc6c8d8b
commit 341e91441a
19 changed files with 412 additions and 105 deletions
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4
webrtc/modules/rtp_rtcp/source/rtp_utility.cc
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@ -391,9 +391,11 @@ bool RTPHeaderParser::Parse(RTPHeader& header,
// If in effect, MAY be omitted for those packets for which the offset
// is zero.
header.extension.hasTransmissionTimeOffset = false;
header.extension.transmissionTimeOffset = 0;
// May not be present in packet.
header.extension.hasAbsoluteSendTime = false;
header.extension.absoluteSendTime = 0;
if (X) {
@ -490,6 +492,7 @@ void RTPHeaderParser::ParseOneByteExtensionHeader(
// Negative offset, correct sign for Word24 to Word32.
header.extension.transmissionTimeOffset |= 0xFF000000;
}
header.extension.hasTransmissionTimeOffset = true;
break;
}
case kRtpExtensionAudioLevel: {
@ -524,6 +527,7 @@ void RTPHeaderParser::ParseOneByteExtensionHeader(
absoluteSendTime += *ptr++ << 8;
absoluteSendTime += *ptr++;
header.extension.absoluteSendTime = absoluteSendTime;
header.extension.hasAbsoluteSendTime = true;
break;
}
default: {