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 98c8c5dbc09dfb50fd891e7528bff1e2d2ca42a2..0000000000000000000000000000000000000000 |
--- a/webrtc/video/vie_receiver.cc |
+++ /dev/null |
@@ -1,482 +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/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/remote_ntp_time_estimator.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_payload_registry.h" |
-#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h" |
-#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" |
-#include "webrtc/modules/video_coding/include/video_coding.h" |
-#include "webrtc/system_wrappers/include/critical_section_wrapper.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 { |
- |
-static const int kPacketLogIntervalMs = 10000; |
- |
-ViEReceiver::ViEReceiver(VideoCodingModule* module_vcm, |
- RemoteBitrateEstimator* remote_bitrate_estimator, |
- RtpFeedback* rtp_feedback) |
- : receive_cs_(CriticalSectionWrapper::CreateCriticalSection()), |
- clock_(Clock::GetRealTimeClock()), |
- rtp_header_parser_(RtpHeaderParser::Create()), |
- rtp_payload_registry_( |
- new RTPPayloadRegistry(RTPPayloadStrategy::CreateStrategy(false))), |
- rtp_receiver_( |
- RtpReceiver::CreateVideoReceiver(clock_, |
- this, |
- rtp_feedback, |
- rtp_payload_registry_.get())), |
- rtp_receive_statistics_(ReceiveStatistics::Create(clock_)), |
- fec_receiver_(FecReceiver::Create(this)), |
- rtp_rtcp_(NULL), |
- vcm_(module_vcm), |
- remote_bitrate_estimator_(remote_bitrate_estimator), |
- ntp_estimator_(new RemoteNtpTimeEstimator(clock_)), |
- receiving_(false), |
- restored_packet_in_use_(false), |
- receiving_ast_enabled_(false), |
- receiving_cvo_enabled_(false), |
- receiving_tsn_enabled_(false), |
- last_packet_log_ms_(-1) { |
- assert(remote_bitrate_estimator); |
-} |
- |
-ViEReceiver::~ViEReceiver() { |
- UpdateHistograms(); |
-} |
- |
-void ViEReceiver::UpdateHistograms() { |
- FecPacketCounter counter = fec_receiver_->GetPacketCounter(); |
- if (counter.num_packets > 0) { |
- RTC_HISTOGRAM_PERCENTAGE( |
- "WebRTC.Video.ReceivedFecPacketsInPercent", |
- static_cast<int>(counter.num_fec_packets * 100 / counter.num_packets)); |
- } |
- if (counter.num_fec_packets > 0) { |
- RTC_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 RegisterPayload(video_codec); |
-} |
- |
-bool ViEReceiver::RegisterPayload(const VideoCodec& video_codec) { |
- return rtp_receiver_->RegisterReceivePayload(video_codec.plName, |
- video_codec.plType, |
- kVideoPayloadTypeFrequency, |
- 0, |
- video_codec.maxBitrate) == 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); |
-} |
- |
-void ViEReceiver::SetRtpRtcpModule(RtpRtcp* module) { |
- rtp_rtcp_ = module; |
-} |
- |
-RtpReceiver* ViEReceiver::GetRtpReceiver() const { |
- return rtp_receiver_.get(); |
-} |
- |
-void ViEReceiver::RegisterRtpRtcpModules( |
- const std::vector<RtpRtcp*>& rtp_modules) { |
- CriticalSectionScoped cs(receive_cs_.get()); |
- // Only change the "simulcast" modules, the base module can be accessed |
- // without a lock whereas the simulcast modules require locking as they can be |
- // changed in runtime. |
- rtp_rtcp_simulcast_ = |
- std::vector<RtpRtcp*>(rtp_modules.begin() + 1, rtp_modules.end()); |
-} |
- |
-bool ViEReceiver::SetReceiveTimestampOffsetStatus(bool enable, int id) { |
- if (enable) { |
- return rtp_header_parser_->RegisterRtpHeaderExtension( |
- kRtpExtensionTransmissionTimeOffset, id); |
- } else { |
- return rtp_header_parser_->DeregisterRtpHeaderExtension( |
- kRtpExtensionTransmissionTimeOffset); |
- } |
-} |
- |
-bool ViEReceiver::SetReceiveAbsoluteSendTimeStatus(bool enable, int id) { |
- if (enable) { |
- if (rtp_header_parser_->RegisterRtpHeaderExtension( |
- kRtpExtensionAbsoluteSendTime, id)) { |
- receiving_ast_enabled_ = true; |
- return true; |
- } else { |
- return false; |
- } |
- } else { |
- receiving_ast_enabled_ = false; |
- return rtp_header_parser_->DeregisterRtpHeaderExtension( |
- kRtpExtensionAbsoluteSendTime); |
- } |
-} |
- |
-bool ViEReceiver::SetReceiveVideoRotationStatus(bool enable, int id) { |
- if (enable) { |
- if (rtp_header_parser_->RegisterRtpHeaderExtension( |
- kRtpExtensionVideoRotation, id)) { |
- receiving_cvo_enabled_ = true; |
- return true; |
- } else { |
- return false; |
- } |
- } else { |
- receiving_cvo_enabled_ = false; |
- return rtp_header_parser_->DeregisterRtpHeaderExtension( |
- kRtpExtensionVideoRotation); |
- } |
-} |
- |
-bool ViEReceiver::SetReceiveTransportSequenceNumber(bool enable, int id) { |
- if (enable) { |
- if (rtp_header_parser_->RegisterRtpHeaderExtension( |
- kRtpExtensionTransportSequenceNumber, id)) { |
- receiving_tsn_enabled_ = true; |
- return true; |
- } else { |
- return false; |
- } |
- } else { |
- receiving_tsn_enabled_ = false; |
- return rtp_header_parser_->DeregisterRtpHeaderExtension( |
- kRtpExtensionTransportSequenceNumber); |
- } |
-} |
- |
-int ViEReceiver::ReceivedRTPPacket(const void* rtp_packet, |
- size_t rtp_packet_length, |
- const PacketTime& packet_time) { |
- return InsertRTPPacket(static_cast<const uint8_t*>(rtp_packet), |
- rtp_packet_length, packet_time); |
-} |
- |
-int ViEReceiver::ReceivedRTCPPacket(const void* rtcp_packet, |
- size_t rtcp_packet_length) { |
- return InsertRTCPPacket(static_cast<const uint8_t*>(rtcp_packet), |
- rtcp_packet_length); |
-} |
- |
-int32_t ViEReceiver::OnReceivedPayloadData(const uint8_t* payload_data, |
- const size_t payload_size, |
- const WebRtcRTPHeader* rtp_header) { |
- WebRtcRTPHeader rtp_header_with_ntp = *rtp_header; |
- rtp_header_with_ntp.ntp_time_ms = |
- ntp_estimator_->Estimate(rtp_header->header.timestamp); |
- if (vcm_->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); |
-} |
- |
-int ViEReceiver::InsertRTPPacket(const uint8_t* rtp_packet, |
- size_t rtp_packet_length, |
- const PacketTime& packet_time) { |
- { |
- CriticalSectionScoped cs(receive_cs_.get()); |
- if (!receiving_) { |
- return -1; |
- } |
- } |
- |
- RTPHeader header; |
- if (!rtp_header_parser_->Parse(rtp_packet, rtp_packet_length, |
- &header)) { |
- return -1; |
- } |
- 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. |
- CriticalSectionScoped cs(receive_cs_.get()); |
- 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); |
- int ret = ReceivePacket(rtp_packet, rtp_packet_length, header, in_order) |
- ? 0 |
- : -1; |
- // 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 vcm 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; |
- CriticalSectionScoped cs(receive_cs_.get()); |
- 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"; |
- 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(NULL, 0, &rtp_header); |
-} |
- |
-int ViEReceiver::InsertRTCPPacket(const uint8_t* rtcp_packet, |
- size_t rtcp_packet_length) { |
- { |
- CriticalSectionScoped cs(receive_cs_.get()); |
- if (!receiving_) { |
- return -1; |
- } |
- |
- for (RtpRtcp* rtp_rtcp : rtp_rtcp_simulcast_) |
- rtp_rtcp->IncomingRtcpPacket(rtcp_packet, rtcp_packet_length); |
- } |
- assert(rtp_rtcp_); // Should be set by owner at construction time. |
- int ret = rtp_rtcp_->IncomingRtcpPacket(rtcp_packet, rtcp_packet_length); |
- if (ret != 0) { |
- return ret; |
- } |
- |
- int64_t rtt = 0; |
- rtp_rtcp_->RTT(rtp_receiver_->SSRC(), &rtt, NULL, NULL, NULL); |
- if (rtt == 0) { |
- // Waiting for valid rtt. |
- return 0; |
- } |
- uint32_t ntp_secs = 0; |
- uint32_t ntp_frac = 0; |
- uint32_t rtp_timestamp = 0; |
- if (0 != rtp_rtcp_->RemoteNTP(&ntp_secs, &ntp_frac, NULL, NULL, |
- &rtp_timestamp)) { |
- // Waiting for RTCP. |
- return 0; |
- } |
- ntp_estimator_->UpdateRtcpTimestamp(rtt, ntp_secs, ntp_frac, rtp_timestamp); |
- |
- return 0; |
-} |
- |
-void ViEReceiver::StartReceive() { |
- CriticalSectionScoped cs(receive_cs_.get()); |
- receiving_ = true; |
-} |
- |
-void ViEReceiver::StopReceive() { |
- CriticalSectionScoped cs(receive_cs_.get()); |
- 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(), NULL, NULL, &min_rtt, NULL); |
- return !in_order && |
- statistician->IsRetransmitOfOldPacket(header, min_rtt); |
-} |
-} // namespace webrtc |