Index: webrtc/video/vie_receiver.cc |
diff --git a/webrtc/video/vie_receiver.cc b/webrtc/video/vie_receiver.cc |
deleted file mode 100644 |
index 95d2f6fc8e3e3ec84486357b2e40a7a3402c0960..0000000000000000000000000000000000000000 |
--- a/webrtc/video/vie_receiver.cc |
+++ /dev/null |
@@ -1,440 +0,0 @@ |
-/* |
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
- * |
- * Use of this source code is governed by a BSD-style license |
- * that can be found in the LICENSE file in the root of the source |
- * tree. An additional intellectual property rights grant can be found |
- * in the file PATENTS. All contributing project authors may |
- * be found in the AUTHORS file in the root of the source tree. |
- */ |
- |
-#include "webrtc/video/vie_receiver.h" |
- |
-#include <vector> |
- |
-#include "webrtc/base/logging.h" |
-#include "webrtc/config.h" |
-#include "webrtc/modules/pacing/packet_router.h" |
-#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" |
-#include "webrtc/modules/rtp_rtcp/include/fec_receiver.h" |
-#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h" |
-#include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h" |
-#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" |
-#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h" |
-#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" |
-#include "webrtc/modules/video_coding/video_coding_impl.h" |
-#include "webrtc/system_wrappers/include/metrics.h" |
-#include "webrtc/system_wrappers/include/tick_util.h" |
-#include "webrtc/system_wrappers/include/timestamp_extrapolator.h" |
-#include "webrtc/system_wrappers/include/trace.h" |
- |
-namespace webrtc { |
- |
-std::unique_ptr<RtpRtcp> CreateRtpRtcpModule( |
- ReceiveStatistics* receive_statistics, |
- Transport* outgoing_transport, |
- RtcpRttStats* rtt_stats, |
- RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer, |
- RemoteBitrateEstimator* remote_bitrate_estimator, |
- RtpPacketSender* paced_sender, |
- TransportSequenceNumberAllocator* transport_sequence_number_allocator) { |
- RtpRtcp::Configuration configuration; |
- configuration.audio = false; |
- configuration.receiver_only = true; |
- configuration.receive_statistics = receive_statistics; |
- configuration.outgoing_transport = outgoing_transport; |
- configuration.intra_frame_callback = nullptr; |
- configuration.rtt_stats = rtt_stats; |
- configuration.rtcp_packet_type_counter_observer = |
- rtcp_packet_type_counter_observer; |
- configuration.paced_sender = paced_sender; |
- configuration.transport_sequence_number_allocator = |
- transport_sequence_number_allocator; |
- configuration.send_bitrate_observer = nullptr; |
- configuration.send_frame_count_observer = nullptr; |
- configuration.send_side_delay_observer = nullptr; |
- configuration.bandwidth_callback = nullptr; |
- configuration.transport_feedback_callback = nullptr; |
- |
- std::unique_ptr<RtpRtcp> rtp_rtcp(RtpRtcp::CreateRtpRtcp(configuration)); |
- rtp_rtcp->SetSendingStatus(false); |
- rtp_rtcp->SetSendingMediaStatus(false); |
- rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound); |
- |
- return rtp_rtcp; |
-} |
- |
- |
-static const int kPacketLogIntervalMs = 10000; |
- |
-ViEReceiver::ViEReceiver(vcm::VideoReceiver* video_receiver, |
- RemoteBitrateEstimator* remote_bitrate_estimator, |
- RtpFeedback* rtp_feedback, |
- Transport* transport, |
- RtcpRttStats* rtt_stats, |
- PacedSender* paced_sender, |
- PacketRouter* packet_router) |
- : clock_(Clock::GetRealTimeClock()), |
- video_receiver_(video_receiver), |
- remote_bitrate_estimator_(remote_bitrate_estimator), |
- packet_router_(packet_router), |
- ntp_estimator_(clock_), |
- rtp_payload_registry_(RTPPayloadStrategy::CreateStrategy(false)), |
- rtp_header_parser_(RtpHeaderParser::Create()), |
- rtp_receiver_(RtpReceiver::CreateVideoReceiver(clock_, |
- this, |
- rtp_feedback, |
- &rtp_payload_registry_)), |
- rtp_receive_statistics_(ReceiveStatistics::Create(clock_)), |
- fec_receiver_(FecReceiver::Create(this)), |
- receiving_(false), |
- restored_packet_in_use_(false), |
- last_packet_log_ms_(-1), |
- rtp_rtcp_(CreateRtpRtcpModule(rtp_receive_statistics_.get(), |
- transport, |
- rtt_stats, |
- &rtcp_packet_type_counter_observer_, |
- remote_bitrate_estimator_, |
- paced_sender, |
- packet_router)) { |
- packet_router_->AddRtpModule(rtp_rtcp_.get()); |
- rtp_rtcp_->SetKeyFrameRequestMethod(kKeyFrameReqPliRtcp); |
-} |
- |
-ViEReceiver::~ViEReceiver() { |
- packet_router_->RemoveRtpModule(rtp_rtcp_.get()); |
- UpdateHistograms(); |
-} |
- |
-void ViEReceiver::UpdateHistograms() { |
- FecPacketCounter counter = fec_receiver_->GetPacketCounter(); |
- if (counter.num_packets > 0) { |
- RTC_LOGGED_HISTOGRAM_PERCENTAGE( |
- "WebRTC.Video.ReceivedFecPacketsInPercent", |
- static_cast<int>(counter.num_fec_packets * 100 / counter.num_packets)); |
- } |
- if (counter.num_fec_packets > 0) { |
- RTC_LOGGED_HISTOGRAM_PERCENTAGE( |
- "WebRTC.Video.RecoveredMediaPacketsInPercentOfFec", |
- static_cast<int>(counter.num_recovered_packets * 100 / |
- counter.num_fec_packets)); |
- } |
-} |
- |
-bool ViEReceiver::SetReceiveCodec(const VideoCodec& video_codec) { |
- int8_t old_pltype = -1; |
- if (rtp_payload_registry_.ReceivePayloadType( |
- video_codec.plName, kVideoPayloadTypeFrequency, 0, |
- video_codec.maxBitrate, &old_pltype) != -1) { |
- rtp_payload_registry_.DeRegisterReceivePayload(old_pltype); |
- } |
- |
- return rtp_receiver_->RegisterReceivePayload( |
- video_codec.plName, video_codec.plType, kVideoPayloadTypeFrequency, |
- 0, 0) == 0; |
-} |
- |
-void ViEReceiver::SetNackStatus(bool enable, |
- int max_nack_reordering_threshold) { |
- if (!enable) { |
- // Reset the threshold back to the lower default threshold when NACK is |
- // disabled since we no longer will be receiving retransmissions. |
- max_nack_reordering_threshold = kDefaultMaxReorderingThreshold; |
- } |
- rtp_receive_statistics_->SetMaxReorderingThreshold( |
- max_nack_reordering_threshold); |
- rtp_receiver_->SetNACKStatus(enable ? kNackRtcp : kNackOff); |
-} |
- |
-void ViEReceiver::SetRtxPayloadType(int payload_type, |
- int associated_payload_type) { |
- rtp_payload_registry_.SetRtxPayloadType(payload_type, |
- associated_payload_type); |
-} |
- |
-void ViEReceiver::SetUseRtxPayloadMappingOnRestore(bool val) { |
- rtp_payload_registry_.set_use_rtx_payload_mapping_on_restore(val); |
-} |
- |
-void ViEReceiver::SetRtxSsrc(uint32_t ssrc) { |
- rtp_payload_registry_.SetRtxSsrc(ssrc); |
-} |
- |
-bool ViEReceiver::GetRtxSsrc(uint32_t* ssrc) const { |
- return rtp_payload_registry_.GetRtxSsrc(ssrc); |
-} |
- |
-bool ViEReceiver::IsFecEnabled() const { |
- return rtp_payload_registry_.ulpfec_payload_type() > -1; |
-} |
- |
-uint32_t ViEReceiver::GetRemoteSsrc() const { |
- return rtp_receiver_->SSRC(); |
-} |
- |
-int ViEReceiver::GetCsrcs(uint32_t* csrcs) const { |
- return rtp_receiver_->CSRCs(csrcs); |
-} |
- |
-RtpReceiver* ViEReceiver::GetRtpReceiver() const { |
- return rtp_receiver_.get(); |
-} |
- |
-void ViEReceiver::EnableReceiveRtpHeaderExtension(const std::string& extension, |
- int id) { |
- RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension)); |
- RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension( |
- StringToRtpExtensionType(extension), id)); |
-} |
- |
-void ViEReceiver::RegisterRtcpPacketTypeCounterObserver( |
- RtcpPacketTypeCounterObserver* observer) { |
- rtcp_packet_type_counter_observer_.Set(observer); |
-} |
- |
- |
-int32_t ViEReceiver::OnReceivedPayloadData(const uint8_t* payload_data, |
- const size_t payload_size, |
- const WebRtcRTPHeader* rtp_header) { |
- RTC_DCHECK(video_receiver_); |
- WebRtcRTPHeader rtp_header_with_ntp = *rtp_header; |
- rtp_header_with_ntp.ntp_time_ms = |
- ntp_estimator_.Estimate(rtp_header->header.timestamp); |
- if (video_receiver_->IncomingPacket(payload_data, payload_size, |
- rtp_header_with_ntp) != 0) { |
- // Check this... |
- return -1; |
- } |
- return 0; |
-} |
- |
-bool ViEReceiver::OnRecoveredPacket(const uint8_t* rtp_packet, |
- size_t rtp_packet_length) { |
- RTPHeader header; |
- if (!rtp_header_parser_->Parse(rtp_packet, rtp_packet_length, &header)) { |
- return false; |
- } |
- header.payload_type_frequency = kVideoPayloadTypeFrequency; |
- bool in_order = IsPacketInOrder(header); |
- return ReceivePacket(rtp_packet, rtp_packet_length, header, in_order); |
-} |
- |
-bool ViEReceiver::DeliverRtp(const uint8_t* rtp_packet, |
- size_t rtp_packet_length, |
- const PacketTime& packet_time) { |
- RTC_DCHECK(remote_bitrate_estimator_); |
- { |
- rtc::CritScope lock(&receive_cs_); |
- if (!receiving_) { |
- return false; |
- } |
- } |
- |
- RTPHeader header; |
- if (!rtp_header_parser_->Parse(rtp_packet, rtp_packet_length, |
- &header)) { |
- return false; |
- } |
- size_t payload_length = rtp_packet_length - header.headerLength; |
- int64_t arrival_time_ms; |
- int64_t now_ms = clock_->TimeInMilliseconds(); |
- if (packet_time.timestamp != -1) |
- arrival_time_ms = (packet_time.timestamp + 500) / 1000; |
- else |
- arrival_time_ms = now_ms; |
- |
- { |
- // Periodically log the RTP header of incoming packets. |
- rtc::CritScope lock(&receive_cs_); |
- if (now_ms - last_packet_log_ms_ > kPacketLogIntervalMs) { |
- std::stringstream ss; |
- ss << "Packet received on SSRC: " << header.ssrc << " with payload type: " |
- << static_cast<int>(header.payloadType) << ", timestamp: " |
- << header.timestamp << ", sequence number: " << header.sequenceNumber |
- << ", arrival time: " << arrival_time_ms; |
- if (header.extension.hasTransmissionTimeOffset) |
- ss << ", toffset: " << header.extension.transmissionTimeOffset; |
- if (header.extension.hasAbsoluteSendTime) |
- ss << ", abs send time: " << header.extension.absoluteSendTime; |
- LOG(LS_INFO) << ss.str(); |
- last_packet_log_ms_ = now_ms; |
- } |
- } |
- |
- remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_length, |
- header, true); |
- header.payload_type_frequency = kVideoPayloadTypeFrequency; |
- |
- bool in_order = IsPacketInOrder(header); |
- rtp_payload_registry_.SetIncomingPayloadType(header); |
- bool ret = ReceivePacket(rtp_packet, rtp_packet_length, header, in_order); |
- // Update receive statistics after ReceivePacket. |
- // Receive statistics will be reset if the payload type changes (make sure |
- // that the first packet is included in the stats). |
- rtp_receive_statistics_->IncomingPacket( |
- header, rtp_packet_length, IsPacketRetransmitted(header, in_order)); |
- return ret; |
-} |
- |
-bool ViEReceiver::ReceivePacket(const uint8_t* packet, |
- size_t packet_length, |
- const RTPHeader& header, |
- bool in_order) { |
- if (rtp_payload_registry_.IsEncapsulated(header)) { |
- return ParseAndHandleEncapsulatingHeader(packet, packet_length, header); |
- } |
- const uint8_t* payload = packet + header.headerLength; |
- assert(packet_length >= header.headerLength); |
- size_t payload_length = packet_length - header.headerLength; |
- PayloadUnion payload_specific; |
- if (!rtp_payload_registry_.GetPayloadSpecifics(header.payloadType, |
- &payload_specific)) { |
- return false; |
- } |
- return rtp_receiver_->IncomingRtpPacket(header, payload, payload_length, |
- payload_specific, in_order); |
-} |
- |
-bool ViEReceiver::ParseAndHandleEncapsulatingHeader(const uint8_t* packet, |
- size_t packet_length, |
- const RTPHeader& header) { |
- if (rtp_payload_registry_.IsRed(header)) { |
- int8_t ulpfec_pt = rtp_payload_registry_.ulpfec_payload_type(); |
- if (packet[header.headerLength] == ulpfec_pt) { |
- rtp_receive_statistics_->FecPacketReceived(header, packet_length); |
- // Notify video_receiver about received FEC packets to avoid NACKing these |
- // packets. |
- NotifyReceiverOfFecPacket(header); |
- } |
- if (fec_receiver_->AddReceivedRedPacket( |
- header, packet, packet_length, ulpfec_pt) != 0) { |
- return false; |
- } |
- return fec_receiver_->ProcessReceivedFec() == 0; |
- } else if (rtp_payload_registry_.IsRtx(header)) { |
- if (header.headerLength + header.paddingLength == packet_length) { |
- // This is an empty packet and should be silently dropped before trying to |
- // parse the RTX header. |
- return true; |
- } |
- // Remove the RTX header and parse the original RTP header. |
- if (packet_length < header.headerLength) |
- return false; |
- if (packet_length > sizeof(restored_packet_)) |
- return false; |
- rtc::CritScope lock(&receive_cs_); |
- if (restored_packet_in_use_) { |
- LOG(LS_WARNING) << "Multiple RTX headers detected, dropping packet."; |
- return false; |
- } |
- if (!rtp_payload_registry_.RestoreOriginalPacket( |
- restored_packet_, packet, &packet_length, rtp_receiver_->SSRC(), |
- header)) { |
- LOG(LS_WARNING) << "Incoming RTX packet: Invalid RTP header ssrc: " |
- << header.ssrc << " payload type: " |
- << static_cast<int>(header.payloadType); |
- return false; |
- } |
- restored_packet_in_use_ = true; |
- bool ret = OnRecoveredPacket(restored_packet_, packet_length); |
- restored_packet_in_use_ = false; |
- return ret; |
- } |
- return false; |
-} |
- |
-void ViEReceiver::NotifyReceiverOfFecPacket(const RTPHeader& header) { |
- int8_t last_media_payload_type = |
- rtp_payload_registry_.last_received_media_payload_type(); |
- if (last_media_payload_type < 0) { |
- LOG(LS_WARNING) << "Failed to get last media payload type."; |
- return; |
- } |
- // Fake an empty media packet. |
- WebRtcRTPHeader rtp_header = {}; |
- rtp_header.header = header; |
- rtp_header.header.payloadType = last_media_payload_type; |
- rtp_header.header.paddingLength = 0; |
- PayloadUnion payload_specific; |
- if (!rtp_payload_registry_.GetPayloadSpecifics(last_media_payload_type, |
- &payload_specific)) { |
- LOG(LS_WARNING) << "Failed to get payload specifics."; |
- return; |
- } |
- rtp_header.type.Video.codec = payload_specific.Video.videoCodecType; |
- rtp_header.type.Video.rotation = kVideoRotation_0; |
- if (header.extension.hasVideoRotation) { |
- rtp_header.type.Video.rotation = |
- ConvertCVOByteToVideoRotation(header.extension.videoRotation); |
- } |
- OnReceivedPayloadData(nullptr, 0, &rtp_header); |
-} |
- |
-bool ViEReceiver::DeliverRtcp(const uint8_t* rtcp_packet, |
- size_t rtcp_packet_length) { |
- { |
- rtc::CritScope lock(&receive_cs_); |
- if (!receiving_) { |
- return false; |
- } |
- } |
- |
- rtp_rtcp_->IncomingRtcpPacket(rtcp_packet, rtcp_packet_length); |
- |
- int64_t rtt = 0; |
- rtp_rtcp_->RTT(rtp_receiver_->SSRC(), &rtt, nullptr, nullptr, nullptr); |
- if (rtt == 0) { |
- // Waiting for valid rtt. |
- return true; |
- } |
- uint32_t ntp_secs = 0; |
- uint32_t ntp_frac = 0; |
- uint32_t rtp_timestamp = 0; |
- if (rtp_rtcp_->RemoteNTP(&ntp_secs, &ntp_frac, nullptr, nullptr, |
- &rtp_timestamp) != 0) { |
- // Waiting for RTCP. |
- return true; |
- } |
- ntp_estimator_.UpdateRtcpTimestamp(rtt, ntp_secs, ntp_frac, rtp_timestamp); |
- |
- return true; |
-} |
- |
-void ViEReceiver::StartReceive() { |
- rtc::CritScope lock(&receive_cs_); |
- receiving_ = true; |
-} |
- |
-void ViEReceiver::StopReceive() { |
- rtc::CritScope lock(&receive_cs_); |
- receiving_ = false; |
-} |
- |
-ReceiveStatistics* ViEReceiver::GetReceiveStatistics() const { |
- return rtp_receive_statistics_.get(); |
-} |
- |
-bool ViEReceiver::IsPacketInOrder(const RTPHeader& header) const { |
- StreamStatistician* statistician = |
- rtp_receive_statistics_->GetStatistician(header.ssrc); |
- if (!statistician) |
- return false; |
- return statistician->IsPacketInOrder(header.sequenceNumber); |
-} |
- |
-bool ViEReceiver::IsPacketRetransmitted(const RTPHeader& header, |
- bool in_order) const { |
- // Retransmissions are handled separately if RTX is enabled. |
- if (rtp_payload_registry_.RtxEnabled()) |
- return false; |
- StreamStatistician* statistician = |
- rtp_receive_statistics_->GetStatistician(header.ssrc); |
- if (!statistician) |
- return false; |
- // Check if this is a retransmission. |
- int64_t min_rtt = 0; |
- rtp_rtcp_->RTT(rtp_receiver_->SSRC(), nullptr, nullptr, &min_rtt, nullptr); |
- return !in_order && |
- statistician->IsRetransmitOfOldPacket(header, min_rtt); |
-} |
-} // namespace webrtc |