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Move Ownership of RtpModules to VideoSendStream from VieChannel and remove use of vie_channel and vie_receiver from video_send_stream.

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The purpose of this refactoring is a first step of separating the encoder parts from the RTP transport.

BUG=webrtc:5687
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1864313003 .

Cr-Commit-Position: refs/heads/master@{#12377}
This commit is contained in:
Per
2016-04-15 14:59:13 +02:00
parent 6ca0a31708
commit 83d0910694
17 changed files with 360 additions and 543 deletions
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35
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
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@ -300,30 +300,21 @@ void ModuleRtpRtcpImpl::SetSequenceNumber(const uint16_t seq_num) {
rtp_sender_.SetSequenceNumber(seq_num);
}
bool ModuleRtpRtcpImpl::SetRtpStateForSsrc(uint32_t ssrc,
const RtpState& rtp_state) {
if (rtp_sender_.SSRC() == ssrc) {
SetStartTimestamp(rtp_state.start_timestamp);
rtp_sender_.SetRtpState(rtp_state);
return true;
}
if (rtp_sender_.RtxSsrc() == ssrc) {
rtp_sender_.SetRtxRtpState(rtp_state);
return true;
}
return false;
void ModuleRtpRtcpImpl::SetRtpState(const RtpState& rtp_state) {
SetStartTimestamp(rtp_state.start_timestamp);
rtp_sender_.SetRtpState(rtp_state);
}
bool ModuleRtpRtcpImpl::GetRtpStateForSsrc(uint32_t ssrc, RtpState* rtp_state) {
if (rtp_sender_.SSRC() == ssrc) {
*rtp_state = rtp_sender_.GetRtpState();
return true;
}
if (rtp_sender_.RtxSsrc() == ssrc) {
*rtp_state = rtp_sender_.GetRtxRtpState();
return true;
}
return false;
void ModuleRtpRtcpImpl::SetRtxState(const RtpState& rtp_state) {
rtp_sender_.SetRtxRtpState(rtp_state);
}
RtpState ModuleRtpRtcpImpl::GetRtpState() const {
return rtp_sender_.GetRtpState();
}
RtpState ModuleRtpRtcpImpl::GetRtxState() const {
return rtp_sender_.GetRtxRtpState();
}
uint32_t ModuleRtpRtcpImpl::SSRC() const {