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Unified Diff: webrtc/voice_engine/channel.cc

Issue 1705763002: Remove PacketRouter sender distinction. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 10 months ago
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Index: webrtc/voice_engine/channel.cc
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index c078d20a91cc513b758206de427f8a991f3810b2..b3fff37bdaed489557413c0d6a683fcc6ccd56a0 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -855,6 +855,7 @@ Channel::Channel(int32_t channelId,
configuration.event_log = event_log;
_rtpRtcpModule.reset(RtpRtcp::CreateRtpRtcp(configuration));
+ _rtpRtcpModule->SetSendingMediaStatus(false);
the sun 2016/02/17 19:54:54 Now that both voe/vie channels set "sending media"
pbos-webrtc 2016/02/18 13:09:42 Done.
statistics_proxy_.reset(new StatisticsProxy(_rtpRtcpModule->SSRC()));
rtp_receive_statistics_->RegisterRtcpStatisticsCallback(
@@ -1136,10 +1137,12 @@ int32_t Channel::StartSend() {
}
channel_state_.SetSending(true);
+ _rtpRtcpModule->SetSendingMediaStatus(true);
if (_rtpRtcpModule->SetSendingStatus(true) != 0) {
_engineStatisticsPtr->SetLastError(
VE_RTP_RTCP_MODULE_ERROR, kTraceError,
"StartSend() RTP/RTCP failed to start sending");
+ _rtpRtcpModule->SetSendingMediaStatus(false);
rtc::CritScope cs(&_callbackCritSect);
channel_state_.SetSending(false);
return -1;
@@ -1171,6 +1174,7 @@ int32_t Channel::StopSend() {
VE_RTP_RTCP_MODULE_ERROR, kTraceWarning,
"StartSend() RTP/RTCP failed to stop sending");
}
+ _rtpRtcpModule->SetSendingMediaStatus(false);
return 0;
}
@@ -2599,14 +2603,14 @@ void Channel::RegisterSenderCongestionControlObjects(
seq_num_allocator_proxy_->SetSequenceNumberAllocator(packet_router);
rtp_packet_sender_proxy_->SetPacketSender(rtp_packet_sender);
_rtpRtcpModule->SetStorePacketsStatus(true, 600);
- packet_router->AddRtpModule(_rtpRtcpModule.get(), true);
+ packet_router->AddRtpModule(_rtpRtcpModule.get());
packet_router_ = packet_router;
}
void Channel::RegisterReceiverCongestionControlObjects(
PacketRouter* packet_router) {
RTC_DCHECK(packet_router && !packet_router_);
- packet_router->AddRtpModule(_rtpRtcpModule.get(), false);
+ packet_router->AddRtpModule(_rtpRtcpModule.get());
packet_router_ = packet_router;
}
@@ -2615,8 +2619,7 @@ void Channel::ResetCongestionControlObjects() {
_rtpRtcpModule->SetStorePacketsStatus(false, 600);
feedback_observer_proxy_->SetTransportFeedbackObserver(nullptr);
seq_num_allocator_proxy_->SetSequenceNumberAllocator(nullptr);
- const bool sender = rtp_packet_sender_proxy_->HasPacketSender();
the sun 2016/02/17 19:54:54 Remove HasPacketSender() from RtpPacketSenderProxy
pbos-webrtc 2016/02/18 13:09:42 Done.
- packet_router_->RemoveRtpModule(_rtpRtcpModule.get(), sender);
+ packet_router_->RemoveRtpModule(_rtpRtcpModule.get());
packet_router_ = nullptr;
rtp_packet_sender_proxy_->SetPacketSender(nullptr);
}
« webrtc/modules/pacing/packet_router_unittest.cc ('K') | « webrtc/video/vie_channel.cc ('k') | no next file » | no next file with comments »

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