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pre-factor cleanup pre-work.

Browse Source 
Review URL: https://webrtc-codereview.appspot.com/938010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3054 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pwestin@webrtc.org
2012-11-07 17:01:04 +00:00
parent 4cebe6cded
commit c66e8b3f31
4 changed files with 803 additions and 1111 deletions
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38
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
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@ -52,7 +52,11 @@ RtpRtcp* RtpRtcp::CreateRtpRtcp(const RtpRtcp::Configuration& configuration) {
}
ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
: _rtpSender(configuration.id, configuration.audio, configuration.clock),
: _rtpSender(configuration.id,
configuration.audio,
configuration.clock,
configuration.outgoing_transport,
configuration.audio_messages),
_rtpReceiver(configuration.id, configuration.audio, configuration.clock,
this),
_rtcpSender(configuration.id, configuration.audio, configuration.clock,
@ -97,10 +101,8 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
_rtcpReceiver.RegisterRtcpObservers(configuration.intra_frame_callback,
configuration.bandwidth_callback,
configuration.rtcp_feedback);
_rtpSender.RegisterAudioCallback(configuration.audio_messages);
_rtpReceiver.RegisterIncomingAudioCallback(configuration.audio_messages);
_rtpSender.RegisterSendTransport(configuration.outgoing_transport);
_rtcpSender.RegisterSendTransport(configuration.outgoing_transport);
// make sure that RTCP objects are aware of our SSRC
@ -628,7 +630,8 @@ WebRtc_Word32 ModuleRtpRtcpImpl::SetStartTimestamp(
"SetStartTimestamp(%d)",
timestamp);
_rtcpSender.SetStartTimestamp(timestamp);
return _rtpSender.SetStartTimestamp(timestamp, true);
_rtpSender.SetStartTimestamp(timestamp, true);
return 0; // TODO(pwestin): change to void.
}
WebRtc_UWord16 ModuleRtpRtcpImpl::SequenceNumber() const {
@ -646,7 +649,8 @@ WebRtc_Word32 ModuleRtpRtcpImpl::SetSequenceNumber(
"SetSequenceNumber(%d)",
seqNum);
return _rtpSender.SetSequenceNumber(seqNum);
_rtpSender.SetSequenceNumber(seqNum);
return 0; // TODO(pwestin): change to void.
}
WebRtc_UWord32 ModuleRtpRtcpImpl::SSRC() const {
@ -659,17 +663,16 @@ WebRtc_UWord32 ModuleRtpRtcpImpl::SSRC() const {
WebRtc_Word32 ModuleRtpRtcpImpl::SetSSRC(const WebRtc_UWord32 ssrc) {
WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "SetSSRC(%d)", ssrc);
if (_rtpSender.SetSSRC(ssrc) == 0) {
_rtcpReceiver.SetSSRC(ssrc);
_rtcpSender.SetSSRC(ssrc);
return 0;
}
return -1;
_rtpSender.SetSSRC(ssrc);
_rtcpReceiver.SetSSRC(ssrc);
_rtcpSender.SetSSRC(ssrc);
return 0; // TODO(pwestin): change to void.
}
WebRtc_Word32 ModuleRtpRtcpImpl::SetCSRCStatus(const bool include) {
_rtcpSender.SetCSRCStatus(include);
return _rtpSender.SetCSRCStatus(include);
_rtpSender.SetCSRCStatus(include);
return 0; // TODO(pwestin): change to void.
}
WebRtc_Word32 ModuleRtpRtcpImpl::CSRCs(
@ -702,16 +705,15 @@ WebRtc_Word32 ModuleRtpRtcpImpl::SetCSRCs(
}
it++;
}
return 0;
} else {
for (int i = 0; i < arrLength; i++) {
WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "\tidx:%d CSRC:%u", i,
arrOfCSRC[i]);
}
_rtcpSender.SetCSRCs(arrOfCSRC, arrLength);
return _rtpSender.SetCSRCs(arrOfCSRC, arrLength);
_rtpSender.SetCSRCs(arrOfCSRC, arrLength);
}
return 0; // TODO(pwestin): change to void.
}
WebRtc_UWord32 ModuleRtpRtcpImpl::PacketCountSent() const {
@ -1129,7 +1131,8 @@ WebRtc_Word32 ModuleRtpRtcpImpl::ResetSendDataCountersRTP() {
WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id,
"ResetSendDataCountersRTP()");
return _rtpSender.ResetDataCounters();
_rtpSender.ResetDataCounters();
return 0; // TODO(pwestin): change to void.
}
// Force a send of an RTCP packet
@ -1495,7 +1498,8 @@ WebRtc_Word32 ModuleRtpRtcpImpl::SetStorePacketsStatus(
WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id,
"SetStorePacketsStatus(disable)");
}
return _rtpSender.SetStorePacketsStatus(enable, numberToStore);
_rtpSender.SetStorePacketsStatus(enable, numberToStore);
return 0; // TODO(pwestin): change to void.
}
/*