Index: webrtc/video/video_send_stream.cc |
diff --git a/webrtc/video/video_send_stream.cc b/webrtc/video/video_send_stream.cc |
index 5c3f74fd7e815a7fdf13b194a82130b8698dc2f2..bc141e86afd90f0d904e4e53f1d484729df7e396 100644 |
--- a/webrtc/video/video_send_stream.cc |
+++ b/webrtc/video/video_send_stream.cc |
@@ -169,6 +169,7 @@ VideoSendStream::VideoSendStream( |
config_.rtp.ssrcs.size(), |
true), |
vie_receiver_(vie_channel_.vie_receiver()), |
+ rtp_rtcp_modules_(vie_channel_.rtp_rtcp()), |
input_(&vie_encoder_, |
config_.local_renderer, |
&stats_proxy_, |
@@ -209,27 +210,31 @@ VideoSendStream::VideoSendStream( |
} |
} |
- RtpRtcp* rtp_module = vie_channel_.rtp_rtcp(); |
- remb_->AddRembSender(rtp_module); |
- rtp_module->SetREMBStatus(true); |
+ remb_->AddRembSender(rtp_rtcp_modules_[0]); |
+ rtp_rtcp_modules_[0]->SetREMBStatus(true); |
// Enable NACK, FEC or both. |
const bool enable_protection_nack = config_.rtp.nack.rtp_history_ms > 0; |
const bool enable_protection_fec = config_.rtp.fec.red_payload_type != -1; |
// TODO(changbin): Should set RTX for RED mapping in RTP sender in future. |
vie_channel_.SetProtectionMode(enable_protection_nack, enable_protection_fec, |
- config_.rtp.fec.red_payload_type, |
- config_.rtp.fec.ulpfec_payload_type); |
+ config_.rtp.fec.red_payload_type, |
+ config_.rtp.fec.ulpfec_payload_type); |
vie_encoder_.SetProtectionMethod(enable_protection_nack, |
- enable_protection_fec); |
+ enable_protection_fec); |
ConfigureSsrcs(); |
- vie_channel_.SetRTCPCName(config_.rtp.c_name.c_str()); |
- |
+ // TODO(pbos): Should we set CNAME on all RTP modules? |
+ rtp_rtcp_modules_.front()->SetCNAME(config_.rtp.c_name.c_str()); |
// 28 to match packet overhead in ModuleRtpRtcpImpl. |
RTC_DCHECK_LE(config_.rtp.max_packet_size, static_cast<size_t>(0xFFFF - 28)); |
- vie_channel_.SetMTU(static_cast<uint16_t>(config_.rtp.max_packet_size + 28)); |
+ const uint16_t mtu = static_cast<uint16_t>(config_.rtp.max_packet_size + 28); |
stefan-webrtc
2016/02/22 13:11:52
Make 28 a named constant.
pbos-webrtc
2016/02/22 13:21:01
Done.
|
+ for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
+ rtp_rtcp->RegisterRtcpStatisticsCallback(&stats_proxy_); |
+ rtp_rtcp->RegisterSendChannelRtpStatisticsCallback(&stats_proxy_); |
+ rtp_rtcp->SetMaxTransferUnit(mtu); |
+ } |
RTC_DCHECK(config.encoder_settings.encoder != nullptr); |
RTC_DCHECK_GE(config.encoder_settings.payload_type, 0); |
@@ -251,8 +256,6 @@ VideoSendStream::VideoSendStream( |
encoder_feedback_.AddEncoder(config_.rtp.ssrcs, &vie_encoder_); |
- vie_channel_.RegisterSendChannelRtcpStatisticsCallback(&stats_proxy_); |
- vie_channel_.RegisterSendChannelRtpStatisticsCallback(&stats_proxy_); |
vie_channel_.RegisterRtcpPacketTypeCounterObserver(&stats_proxy_); |
vie_channel_.RegisterSendBitrateObserver(&stats_proxy_); |
vie_channel_.RegisterSendFrameCountObserver(&stats_proxy_); |
@@ -272,17 +275,12 @@ VideoSendStream::~VideoSendStream() { |
vie_channel_.RegisterSendFrameCountObserver(nullptr); |
vie_channel_.RegisterSendBitrateObserver(nullptr); |
vie_channel_.RegisterRtcpPacketTypeCounterObserver(nullptr); |
- vie_channel_.RegisterSendChannelRtpStatisticsCallback(nullptr); |
- vie_channel_.RegisterSendChannelRtcpStatisticsCallback(nullptr); |
- vie_encoder_.DeRegisterExternalEncoder( |
- config_.encoder_settings.payload_type); |
+ vie_encoder_.DeRegisterExternalEncoder(config_.encoder_settings.payload_type); |
call_stats_->DeregisterStatsObserver(vie_channel_.GetStatsObserver()); |
- |
- RtpRtcp* rtp_module = vie_channel_.rtp_rtcp(); |
- rtp_module->SetREMBStatus(false); |
- remb_->RemoveRembSender(rtp_module); |
+ rtp_rtcp_modules_[0]->SetREMBStatus(false); |
+ remb_->RemoveRembSender(rtp_rtcp_modules_[0]); |
// Remove the feedback, stop all encoding threads and processing. This must be |
// done before deleting the channel. |
@@ -485,39 +483,49 @@ void VideoSendStream::NormalUsage() { |
} |
void VideoSendStream::ConfigureSsrcs() { |
- vie_channel_.SetSSRC(config_.rtp.ssrcs.front(), kViEStreamTypeNormal, 0); |
+ // Configure regular SSRCs. |
for (size_t i = 0; i < config_.rtp.ssrcs.size(); ++i) { |
uint32_t ssrc = config_.rtp.ssrcs[i]; |
- vie_channel_.SetSSRC(ssrc, kViEStreamTypeNormal, |
- static_cast<unsigned char>(i)); |
+ RtpRtcp* const rtp_rtcp = rtp_rtcp_modules_[i]; |
+ rtp_rtcp->SetSSRC(ssrc); |
+ |
+ // Restore RTP state if previous existed. |
RtpStateMap::iterator it = suspended_ssrcs_.find(ssrc); |
if (it != suspended_ssrcs_.end()) |
- vie_channel_.SetRtpStateForSsrc(ssrc, it->second); |
+ rtp_rtcp->SetRtpStateForSsrc(ssrc, it->second); |
} |
- if (config_.rtp.rtx.ssrcs.empty()) { |
+ // Set up RTX if available. |
+ if (config_.rtp.rtx.ssrcs.empty()) |
return; |
- } |
- // Set up RTX. |
+ // Configure RTX SSRCs. |
RTC_DCHECK_EQ(config_.rtp.rtx.ssrcs.size(), config_.rtp.ssrcs.size()); |
for (size_t i = 0; i < config_.rtp.rtx.ssrcs.size(); ++i) { |
uint32_t ssrc = config_.rtp.rtx.ssrcs[i]; |
- vie_channel_.SetSSRC(config_.rtp.rtx.ssrcs[i], kViEStreamTypeRtx, |
- static_cast<unsigned char>(i)); |
+ RtpRtcp* const rtp_rtcp = rtp_rtcp_modules_[i]; |
+ rtp_rtcp->SetRtxSsrc(ssrc); |
RtpStateMap::iterator it = suspended_ssrcs_.find(ssrc); |
if (it != suspended_ssrcs_.end()) |
- vie_channel_.SetRtpStateForSsrc(ssrc, it->second); |
+ rtp_rtcp->SetRtpStateForSsrc(ssrc, it->second); |
} |
+ // Configure RTX payload types. |
RTC_DCHECK_GE(config_.rtp.rtx.payload_type, 0); |
- vie_channel_.SetRtxSendPayloadType(config_.rtp.rtx.payload_type, |
- config_.encoder_settings.payload_type); |
+ for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
+ rtp_rtcp->SetRtxSendPayloadType(config_.rtp.rtx.payload_type, |
+ config_.encoder_settings.payload_type); |
+ } |
if (config_.rtp.fec.red_payload_type != -1 && |
config_.rtp.fec.red_rtx_payload_type != -1) { |
- vie_channel_.SetRtxSendPayloadType(config_.rtp.fec.red_rtx_payload_type, |
- config_.rtp.fec.red_payload_type); |
+ for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
+ rtp_rtcp->SetRtxSendPayloadType(config_.rtp.fec.red_rtx_payload_type, |
+ config_.rtp.fec.red_payload_type); |
+ } |
} |
+ // Enable RTX in RTP modules. |
+ for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) |
+ rtp_rtcp->SetRtxSendStatus(kRtxRetransmitted | kRtxRedundantPayloads); |
stefan-webrtc
2016/02/22 13:11:52
I think you can move this to line 524 or 517
pbos-webrtc
2016/02/22 13:21:01
Done.
|
} |
std::map<uint32_t, RtpState> VideoSendStream::GetRtpStates() const { |
@@ -539,11 +547,15 @@ void VideoSendStream::SignalNetworkState(NetworkState state) { |
// When network goes up, enable RTCP status before setting transmission state. |
// When it goes down, disable RTCP afterwards. This ensures that any packets |
// sent due to the network state changed will not be dropped. |
- if (state == kNetworkUp) |
- vie_channel_.SetRTCPMode(config_.rtp.rtcp_mode); |
+ if (state == kNetworkUp) { |
+ for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) |
+ rtp_rtcp->SetRTCPStatus(config_.rtp.rtcp_mode); |
+ } |
vie_encoder_.SetNetworkTransmissionState(state == kNetworkUp); |
- if (state == kNetworkDown) |
- vie_channel_.SetRTCPMode(RtcpMode::kOff); |
+ if (state == kNetworkDown) { |
+ for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) |
+ rtp_rtcp->SetRTCPStatus(RtcpMode::kOff); |
+ } |
} |
int64_t VideoSendStream::GetRtt() const { |
@@ -554,8 +566,8 @@ int64_t VideoSendStream::GetRtt() const { |
uint32_t jitter; |
int64_t rtt_ms; |
if (vie_channel_.GetSendRtcpStatistics(&frac_lost, &cumulative_lost, |
- &extended_max_sequence_number, |
- &jitter, &rtt_ms) == 0) { |
+ &extended_max_sequence_number, &jitter, |
+ &rtt_ms) == 0) { |
return rtt_ms; |
} |
return -1; |