Chromium Code Reviews| Index: webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
| diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
| index 36915c53afeb7fdf9c3a18bf7de53ce4d7f67f84..ba319b5fa339990f7f070695446a9fb4fa59dba0 100644 |
| --- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
| +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
| @@ -41,6 +41,8 @@ constexpr uint16_t kMaxInitRtpSeqNumber = 32767; // 2^15 -1. |
| constexpr uint32_t kTimestampTicksPerMs = 90; |
| constexpr int kBitrateStatisticsWindowMs = 1000; |
| +constexpr size_t kFlexfecPacketsToStoreBeforePacing = 50; |
| + |
| const char* FrameTypeToString(FrameType frame_type) { |
| switch (frame_type) { |
| case kEmptyFrame: |
| @@ -94,6 +96,7 @@ RTPSender::RTPSender( |
| payload_type_map_(), |
| rtp_header_extension_map_(), |
| packet_history_(clock), |
| + flexfec_packet_history_(clock), |
| // Statistics |
| rtp_stats_callback_(nullptr), |
| total_bitrate_sent_(kBitrateStatisticsWindowMs, |
| @@ -127,6 +130,13 @@ RTPSender::RTPSender( |
| // Random start, 16 bits. Can't be 0. |
| sequence_number_rtx_ = random_.Rand(1, kMaxInitRtpSeqNumber); |
| sequence_number_ = random_.Rand(1, kMaxInitRtpSeqNumber); |
| + |
| + // Store FlexFEC packets in the packet history data structure, so they can |
| + // be found when paced. |
| + if (flexfec_sender) { |
| + flexfec_packet_history_.SetStorePacketsStatus( |
| + true, kFlexfecPacketsToStoreBeforePacing); |
| + } |
| } |
| RTPSender::~RTPSender() { |
| @@ -685,13 +695,21 @@ void RTPSender::OnReceivedRtcpReportBlocks( |
| } |
| // Called from pacer when we can send the packet. |
| -bool RTPSender::TimeToSendPacket(uint16_t sequence_number, |
| +bool RTPSender::TimeToSendPacket(uint32_t ssrc, |
| + uint16_t sequence_number, |
| int64_t capture_time_ms, |
| bool retransmission, |
| int probe_cluster_id) { |
| - std::unique_ptr<RtpPacketToSend> packet = |
| - packet_history_.GetPacketAndSetSendTime(sequence_number, 0, |
| - retransmission); |
| + std::unique_ptr<RtpPacketToSend> packet; |
| + rtc::Optional<uint32_t> flexfec_ssrc = FlexfecSsrc(); |
| + if (flexfec_ssrc && ssrc == flexfec_ssrc) { |
| + packet = flexfec_packet_history_.GetPacketAndSetSendTime(sequence_number, 0, |
| + retransmission); |
| + } else { |
|
danilchap
2016/11/10 16:15:13
may be check here ssrc is the media ssrc
then that
brandtr
2016/11/11 10:28:16
Done.
|
| + packet = packet_history_.GetPacketAndSetSendTime(sequence_number, 0, |
| + retransmission); |
| + } |
| + |
| if (!packet) { |
| // Packet cannot be found. Allow sending to continue. |
| return true; |
| @@ -839,11 +857,16 @@ bool RTPSender::SendToNetwork(std::unique_ptr<RtpPacketToSend> packet, |
| if (paced_sender_) { |
| uint16_t seq_no = packet->SequenceNumber(); |
| uint32_t ssrc = packet->Ssrc(); |
| + rtc::Optional<uint32_t> flexfec_ssrc = FlexfecSsrc(); |
| // Correct offset between implementations of millisecond time stamps in |
| // TickTime and Clock. |
| int64_t corrected_time_ms = packet->capture_time_ms() + clock_delta_ms_; |
| size_t payload_length = packet->payload_size(); |
| - packet_history_.PutRtpPacket(std::move(packet), storage, false); |
| + if (flexfec_ssrc && ssrc == flexfec_ssrc) { |
| + flexfec_packet_history_.PutRtpPacket(std::move(packet), storage, false); |
| + } else { |
| + packet_history_.PutRtpPacket(std::move(packet), storage, false); |
| + } |
| paced_sender_->InsertPacket(priority, ssrc, seq_no, corrected_time_ms, |
| payload_length, false); |
| @@ -1089,6 +1112,13 @@ uint32_t RTPSender::SSRC() const { |
| return ssrc_; |
| } |
| +rtc::Optional<uint32_t> RTPSender::FlexfecSsrc() const { |
| + if (video_) { |
| + return video_->FlexfecSsrc(); |
| + } |
| + return rtc::Optional<uint32_t>(); // No value. |
|
danilchap
2016/11/10 16:15:13
not sure if comment is helpful.
brandtr
2016/11/11 10:28:16
Removed.
|
| +} |
| + |
| void RTPSender::SetCsrcs(const std::vector<uint32_t>& csrcs) { |
| assert(csrcs.size() <= kRtpCsrcSize); |
| rtc::CritScope lock(&send_critsect_); |