Reland of Merge webrtc/video_engine/ into webrtc/video/

webrtc/video_engine/vie_receiver.cc

Index: webrtc/video_engine/vie_receiver.cc
diff --git a/webrtc/video_engine/vie_receiver.cc b/webrtc/video_engine/vie_receiver.cc
deleted file mode 100644
index 56dfb28f15ba8c314b525e0b8981d16479e71849..0000000000000000000000000000000000000000
--- a/webrtc/video_engine/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_engine/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