webrtc/video_engine/vie_receiver.cc

/*
 *  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/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "webrtc/modules/rtp_rtcp/interface/fec_receiver.h"
#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
#include "webrtc/modules/utility/interface/rtp_dump.h"
#include "webrtc/modules/video_coding/main/interface/video_coding.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/logging.h"
#include "webrtc/system_wrappers/interface/tick_util.h"
#include "webrtc/system_wrappers/interface/timestamp_extrapolator.h"
#include "webrtc/system_wrappers/interface/trace.h"

namespace webrtc {

ViEReceiver::ViEReceiver(const int32_t channel_id,
                         VideoCodingModule* module_vcm,
                         RemoteBitrateEstimator* remote_bitrate_estimator,
                         RtpFeedback* rtp_feedback)
    : receive_cs_(CriticalSectionWrapper::CreateCriticalSection()),
      channel_id_(channel_id),
      rtp_header_parser_(RtpHeaderParser::Create()),
      rtp_payload_registry_(new RTPPayloadRegistry(
          RTPPayloadStrategy::CreateStrategy(false))),
      rtp_receiver_(RtpReceiver::CreateVideoReceiver(
          channel_id, Clock::GetRealTimeClock(), this, rtp_feedback,
          rtp_payload_registry_.get())),
      rtp_receive_statistics_(ReceiveStatistics::Create(
          Clock::GetRealTimeClock())),
      fec_receiver_(FecReceiver::Create(this)),
      rtp_rtcp_(NULL),
      vcm_(module_vcm),
      remote_bitrate_estimator_(remote_bitrate_estimator),
      clock_(Clock::GetRealTimeClock()),
      ts_extrapolator_(
          new TimestampExtrapolator(clock_->TimeInMilliseconds())),
      rtp_dump_(NULL),
      receiving_(false),
      restored_packet_in_use_(false),
      receiving_ast_enabled_(false) {
  assert(remote_bitrate_estimator);
}

ViEReceiver::~ViEReceiver() {
  if (rtp_dump_) {
    rtp_dump_->Stop();
    RtpDump::DestroyRtpDump(rtp_dump_);
    rtp_dump_ = NULL;
  }
}

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::SetRtxStatus(bool enable, uint32_t ssrc) {
  rtp_payload_registry_->SetRtxStatus(enable, ssrc);
}

void ViEReceiver::SetRtxPayloadType(uint32_t payload_type) {
  rtp_payload_registry_->SetRtxPayloadType(payload_type);
}

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::RegisterSimulcastRtpRtcpModules(
    const std::list<RtpRtcp*>& rtp_modules) {
  CriticalSectionScoped cs(receive_cs_.get());
  rtp_rtcp_simulcast_.clear();

  if (!rtp_modules.empty()) {
    rtp_rtcp_simulcast_.insert(rtp_rtcp_simulcast_.begin(),
                               rtp_modules.begin(),
                               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);
  }
}

int ViEReceiver::ReceivedRTPPacket(const void* rtp_packet,
                                   int 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,
                                    int 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 uint16_t payload_size,
    const WebRtcRTPHeader* rtp_header) {
  WebRtcRTPHeader rtp_header_with_ntp = *rtp_header;
  CalculateCaptureNtpTime(&rtp_header_with_ntp);
  if (vcm_->IncomingPacket(payload_data,
                           payload_size,
                           rtp_header_with_ntp) != 0) {
    // Check this...
    return -1;
  }
  return 0;
}

void ViEReceiver::CalculateCaptureNtpTime(WebRtcRTPHeader* rtp_header) {
  if (rtcp_list_.size() < 2) {
    // We need two RTCP SR reports to calculate NTP.
    return;
  }

  int64_t sender_capture_ntp_ms = 0;
  if (!RtpToNtpMs(rtp_header->header.timestamp,
                  rtcp_list_,
                  &sender_capture_ntp_ms)) {
    return;
  }
  uint32_t timestamp = sender_capture_ntp_ms * 90;
  int64_t receiver_capture_ms =
      ts_extrapolator_->ExtrapolateLocalTime(timestamp);
  int64_t ntp_offset =
      clock_->CurrentNtpInMilliseconds() - clock_->TimeInMilliseconds();
  rtp_header->ntp_time_ms = receiver_capture_ms + ntp_offset;
}

bool ViEReceiver::OnRecoveredPacket(const uint8_t* rtp_packet,
                                    int rtp_packet_length) {
  RTPHeader header;
  if (!rtp_header_parser_->Parse(rtp_packet, rtp_packet_length, &header)) {
    WEBRTC_TRACE(kTraceDebug, webrtc::kTraceVideo, channel_id_,
                 "IncomingPacket invalid RTP header");
    return false;
  }
  header.payload_type_frequency = kVideoPayloadTypeFrequency;
  return ReceivePacket(rtp_packet, rtp_packet_length, header, false);
}

void ViEReceiver::ReceivedBWEPacket(
    int64_t arrival_time_ms, int payload_size, const RTPHeader& header) {
  // Only forward if the incoming packet *and* the channel are both configured
  // to receive absolute sender time. RTP time stamps may have different rates
  // for audio and video and shouldn't be mixed.
  if (header.extension.hasAbsoluteSendTime && receiving_ast_enabled_) {
    remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_size,
                                              header);
  }
}

int ViEReceiver::InsertRTPPacket(const uint8_t* rtp_packet,
                                 int rtp_packet_length,
                                 const PacketTime& packet_time) {
  {
    CriticalSectionScoped cs(receive_cs_.get());
    if (!receiving_) {
      return -1;
    }
    if (rtp_dump_) {
      rtp_dump_->DumpPacket(rtp_packet,
                            static_cast<uint16_t>(rtp_packet_length));
    }
  }

  RTPHeader header;
  if (!rtp_header_parser_->Parse(rtp_packet, rtp_packet_length,
                                 &header)) {
    WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideo, channel_id_,
                 "Incoming packet: Invalid RTP header");
    return -1;
  }
  int payload_length = rtp_packet_length - header.headerLength;
  int64_t arrival_time_ms;
  if (packet_time.timestamp != -1)
    arrival_time_ms = (packet_time.timestamp + 500) / 1000;
  else
    arrival_time_ms = TickTime::MillisecondTimestamp();

  remote_bitrate_estimator_->IncomingPacket(arrival_time_ms,
                                            payload_length, header);
  header.payload_type_frequency = kVideoPayloadTypeFrequency;

  bool in_order = IsPacketInOrder(header);
  rtp_receive_statistics_->IncomingPacket(
      header, rtp_packet_length, IsPacketRetransmitted(header, in_order));
  rtp_payload_registry_->SetIncomingPayloadType(header);
  return ReceivePacket(rtp_packet, rtp_packet_length, header, in_order)
      ? 0
      : -1;
}

bool ViEReceiver::ReceivePacket(const uint8_t* packet,
                                int 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;
  int payload_length = packet_length - header.headerLength;
  assert(payload_length >= 0);
  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,
                                                    int 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.ssrc);
    if (fec_receiver_->AddReceivedRedPacket(
            header, packet, packet_length, ulpfec_pt) != 0) {
      WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideo, channel_id_,
                   "Incoming RED packet error");
      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 > static_cast<int>(sizeof(restored_packet_)))
      return false;
    CriticalSectionScoped cs(receive_cs_.get());
    if (restored_packet_in_use_) {
      WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideo, channel_id_,
                   "Multiple RTX headers detected, dropping packet");
      return false;
    }
    uint8_t* restored_packet_ptr = restored_packet_;
    if (!rtp_payload_registry_->RestoreOriginalPacket(
        &restored_packet_ptr, packet, &packet_length, rtp_receiver_->SSRC(),
        header)) {
      WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideo, channel_id_,
                   "Incoming RTX packet: invalid RTP header");
      return false;
    }
    restored_packet_in_use_ = true;
    bool ret = OnRecoveredPacket(restored_packet_ptr, packet_length);
    restored_packet_in_use_ = false;
    return ret;
  }
  return false;
}

int ViEReceiver::InsertRTCPPacket(const uint8_t* rtcp_packet,
                                  int rtcp_packet_length) {
  {
    CriticalSectionScoped cs(receive_cs_.get());
    if (!receiving_) {
      return -1;
    }

    if (rtp_dump_) {
      rtp_dump_->DumpPacket(
          rtcp_packet, static_cast<uint16_t>(rtcp_packet_length));
    }

    std::list<RtpRtcp*>::iterator it = rtp_rtcp_simulcast_.begin();
    while (it != rtp_rtcp_simulcast_.end()) {
      RtpRtcp* rtp_rtcp = *it++;
      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;
  }

  if (!GetRtcpTimestamp()) {
    LOG(LS_WARNING) << "Failed to retrieve timestamp information from RTCP SR.";
  }

  return 0;
}

bool ViEReceiver::GetRtcpTimestamp() {
  uint16_t rtt = 0;
  rtp_rtcp_->RTT(rtp_receiver_->SSRC(), &rtt, NULL, NULL, NULL);
  if (rtt == 0) {
    // Waiting for valid rtt.
    return true;
  }

  // Update RTCP list
  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)) {
    return false;
  }

  bool new_rtcp_sr = false;
  if (!UpdateRtcpList(ntp_secs,
                      ntp_frac,
                      rtp_timestamp,
                      &rtcp_list_,
                      &new_rtcp_sr)) {
    return false;
  }

  if (!new_rtcp_sr) {
    // No new RTCP SR since last time this function was called.
    return true;
  }

  // Update extrapolator with the new arrival time.
  // The extrapolator assumes the TimeInMilliseconds time.
  int64_t receiver_arrival_time = clock_->TimeInMilliseconds();
  int64_t sender_send_time_ms = Clock::NtpToMs(ntp_secs, ntp_frac);
  int64_t sender_arrival_time_90k = (sender_send_time_ms + rtt / 2) * 90;
  ts_extrapolator_->Update(receiver_arrival_time, sender_arrival_time_90k);
  return true;
}

void ViEReceiver::StartReceive() {
  CriticalSectionScoped cs(receive_cs_.get());
  receiving_ = true;
}

void ViEReceiver::StopReceive() {
  CriticalSectionScoped cs(receive_cs_.get());
  receiving_ = false;
}

int ViEReceiver::StartRTPDump(const char file_nameUTF8[1024]) {
  CriticalSectionScoped cs(receive_cs_.get());
  if (rtp_dump_) {
    // Restart it if it already exists and is started
    rtp_dump_->Stop();
  } else {
    rtp_dump_ = RtpDump::CreateRtpDump();
    if (rtp_dump_ == NULL) {
      WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, channel_id_,
                   "StartRTPDump: Failed to create RTP dump");
      return -1;
    }
  }
  if (rtp_dump_->Start(file_nameUTF8) != 0) {
    RtpDump::DestroyRtpDump(rtp_dump_);
    rtp_dump_ = NULL;
    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, channel_id_,
                 "StartRTPDump: Failed to start RTP dump");
    return -1;
  }
  return 0;
}

int ViEReceiver::StopRTPDump() {
  CriticalSectionScoped cs(receive_cs_.get());
  if (rtp_dump_) {
    if (rtp_dump_->IsActive()) {
      rtp_dump_->Stop();
    } else {
      WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, channel_id_,
                   "StopRTPDump: Dump not active");
    }
    RtpDump::DestroyRtpDump(rtp_dump_);
    rtp_dump_ = NULL;
  } else {
    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, channel_id_,
                 "StopRTPDump: RTP dump not started");
    return -1;
  }
  return 0;
}

void ViEReceiver::GetReceiveBandwidthEstimatorStats(
    ReceiveBandwidthEstimatorStats* output) const {
  remote_bitrate_estimator_->GetStats(output);
}

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.
  uint16_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