| Index: webrtc/video/video_send_stream.cc
|
| diff --git a/webrtc/video/video_send_stream.cc b/webrtc/video/video_send_stream.cc
|
| index db00adf24edc1a92f379df6a6c42d2918f150de6..d3dde5ad9c81730349e10c178b4ef91456e065df 100644
|
| --- a/webrtc/video/video_send_stream.cc
|
| +++ b/webrtc/video/video_send_stream.cc
|
| @@ -197,6 +197,7 @@ VideoSendStream::VideoSendStream(
|
| &payload_router_,
|
| bitrate_allocator),
|
| vcm_(vie_encoder_.vcm()),
|
| + rtp_rtcp_modules_(vie_channel_.rtp_rtcp()),
|
| input_(&vie_encoder_,
|
| config_.local_renderer,
|
| &stats_proxy_,
|
| @@ -236,9 +237,8 @@ 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;
|
| @@ -257,18 +257,25 @@ VideoSendStream::VideoSendStream(
|
| }
|
| // 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));
|
| + static const size_t kRtpPacketSizeOverhead = 28;
|
| + RTC_DCHECK_LE(config_.rtp.max_packet_size, 0xFFFFu + kRtpPacketSizeOverhead);
|
| + const uint16_t mtu = static_cast<uint16_t>(config_.rtp.max_packet_size +
|
| + kRtpPacketSizeOverhead);
|
| + 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);
|
| @@ -288,8 +295,6 @@ VideoSendStream::VideoSendStream(
|
| if (config_.suspend_below_min_bitrate)
|
| vie_encoder_.SuspendBelowMinBitrate();
|
|
|
| - 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_);
|
| @@ -305,17 +310,12 @@ 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]);
|
|
|
| // ViEChannel outlives ViEEncoder so remove encoder from feedback before
|
| // destruction.
|
| @@ -509,38 +509,46 @@ 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);
|
| + rtp_rtcp->SetRtxSendStatus(kRtxRetransmitted | kRtxRedundantPayloads);
|
| + }
|
| 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);
|
| + }
|
| }
|
| }
|
|
|
| @@ -563,11 +571,15 @@ 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);
|
| + }
|
| }
|
|
|
| int VideoSendStream::GetPaddingNeededBps() const {
|
|
|