android-5.1.1_r30/s

// Copyright 2014 The Chromium Authors. All rights reserved.
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.

#include "media/cast/net/rtcp/rtcp.h"

#include "media/cast/cast_config.h"
#include "media/cast/cast_defines.h"
#include "media/cast/cast_environment.h"
#include "media/cast/net/cast_transport_defines.h"
#include "media/cast/net/pacing/paced_sender.h"
#include "media/cast/net/rtcp/rtcp_builder.h"
#include "media/cast/net/rtcp/rtcp_defines.h"
#include "media/cast/net/rtcp/rtcp_utility.h"

using base::TimeDelta;

namespace media {
namespace cast {

static const int32 kStatsHistoryWindowMs = 10000;  // 10 seconds.
// Reject packets that are older than 0.5 seconds older than
// the newest packet we've seen so far. This protect internal
// states from crazy routers. (Based on RRTR)
static const int32 kOutOfOrderMaxAgeMs = 500;

namespace {

// A receiver frame event is identified by frame RTP timestamp, event timestamp
// and event type.
// A receiver packet event is identified by all of the above plus packet id.
// The key format is as follows:
// First uint64:
//   bits 0-11: zeroes (unused).
//   bits 12-15: event type ID.
//   bits 16-31: packet ID if packet event, 0 otherwise.
//   bits 32-63: RTP timestamp.
// Second uint64:
//   bits 0-63: event TimeTicks internal value.
std::pair<uint64, uint64> GetReceiverEventKey(
    uint32 frame_rtp_timestamp,
    const base::TimeTicks& event_timestamp,
    uint8 event_type,
    uint16 packet_id_or_zero) {
  uint64 value1 = event_type;
  value1 <<= 16;
  value1 |= packet_id_or_zero;
  value1 <<= 32;
  value1 |= frame_rtp_timestamp;
  return std::make_pair(
      value1, static_cast<uint64>(event_timestamp.ToInternalValue()));
}

}  // namespace


Rtcp::Rtcp(const RtcpCastMessageCallback& cast_callback,
           const RtcpRttCallback& rtt_callback,
           const RtcpLogMessageCallback& log_callback,
           base::TickClock* clock,
           PacedPacketSender* packet_sender,
           uint32 local_ssrc,
           uint32 remote_ssrc)
    : cast_callback_(cast_callback),
      rtt_callback_(rtt_callback),
      log_callback_(log_callback),
      clock_(clock),
      rtcp_builder_(local_ssrc),
      packet_sender_(packet_sender),
      local_ssrc_(local_ssrc),
      remote_ssrc_(remote_ssrc),
      last_report_truncated_ntp_(0),
      local_clock_ahead_by_(ClockDriftSmoother::GetDefaultTimeConstant()),
      lip_sync_rtp_timestamp_(0),
      lip_sync_ntp_timestamp_(0) {
}

Rtcp::~Rtcp() {}

bool Rtcp::IsRtcpPacket(const uint8* packet, size_t length) {
  if (length < kMinLengthOfRtcp) {
    LOG(ERROR) << "Invalid RTCP packet received.";
    return false;
  }

  uint8 packet_type = packet[1];
  return packet_type >= kPacketTypeLow && packet_type <= kPacketTypeHigh;
}

uint32 Rtcp::GetSsrcOfSender(const uint8* rtcp_buffer, size_t length) {
  if (length < kMinLengthOfRtcp)
    return 0;
  uint32 ssrc_of_sender;
  base::BigEndianReader big_endian_reader(
      reinterpret_cast<const char*>(rtcp_buffer), length);
  big_endian_reader.Skip(4);  // Skip header.
  big_endian_reader.ReadU32(&ssrc_of_sender);
  return ssrc_of_sender;
}

bool Rtcp::IncomingRtcpPacket(const uint8* data, size_t length) {
  // Check if this is a valid RTCP packet.
  if (!IsRtcpPacket(data, length)) {
    VLOG(1) << "Rtcp@" << this << "::IncomingRtcpPacket() -- "
            << "Received an invalid (non-RTCP?) packet.";
    return false;
  }

  // Check if this packet is to us.
  uint32 ssrc_of_sender = GetSsrcOfSender(data, length);
  if (ssrc_of_sender != remote_ssrc_) {
    return false;
  }

  // Parse this packet.
  RtcpParser parser(local_ssrc_, remote_ssrc_);
  base::BigEndianReader reader(reinterpret_cast<const char*>(data), length);
  if (parser.Parse(&reader)) {
    if (parser.has_receiver_reference_time_report()) {
      base::TimeTicks t = ConvertNtpToTimeTicks(
          parser.receiver_reference_time_report().ntp_seconds,
          parser.receiver_reference_time_report().ntp_fraction);
      if (t > largest_seen_timestamp_) {
        largest_seen_timestamp_ = t;
      } else if ((largest_seen_timestamp_ - t).InMilliseconds() >
                 kOutOfOrderMaxAgeMs) {
        // Reject packet, it is too old.
        VLOG(1) << "Rejecting RTCP packet as it is too old ("
                << (largest_seen_timestamp_ - t).InMilliseconds()
                << " ms)";
        return true;
      }

      OnReceivedNtp(parser.receiver_reference_time_report().ntp_seconds,
                    parser.receiver_reference_time_report().ntp_fraction);
    }
    if (parser.has_sender_report()) {
      OnReceivedNtp(parser.sender_report().ntp_seconds,
                    parser.sender_report().ntp_fraction);
      OnReceivedLipSyncInfo(parser.sender_report().rtp_timestamp,
                            parser.sender_report().ntp_seconds,
                            parser.sender_report().ntp_fraction);
    }
    if (parser.has_receiver_log()) {
      if (DedupeReceiverLog(parser.mutable_receiver_log())) {
        OnReceivedReceiverLog(parser.receiver_log());
      }
    }
    if (parser.has_last_report()) {
      OnReceivedDelaySinceLastReport(parser.last_report(),
                                     parser.delay_since_last_report());
    }
    if (parser.has_cast_message()) {
      parser.mutable_cast_message()->ack_frame_id =
          ack_frame_id_wrap_helper_.MapTo32bitsFrameId(
              parser.mutable_cast_message()->ack_frame_id);
      OnReceivedCastFeedback(parser.cast_message());
    }
  }
  return true;
}

bool Rtcp::DedupeReceiverLog(RtcpReceiverLogMessage* receiver_log) {
  RtcpReceiverLogMessage::iterator i = receiver_log->begin();
  while (i != receiver_log->end()) {
    RtcpReceiverEventLogMessages* messages = &i->event_log_messages_;
    RtcpReceiverEventLogMessages::iterator j = messages->begin();
    while (j != messages->end()) {
      ReceiverEventKey key = GetReceiverEventKey(i->rtp_timestamp_,
                                                 j->event_timestamp,
                                                 j->type,
                                                 j->packet_id);
      RtcpReceiverEventLogMessages::iterator tmp = j;
      ++j;
      if (receiver_event_key_set_.insert(key).second) {
        receiver_event_key_queue_.push(key);
        if (receiver_event_key_queue_.size() > kReceiverRtcpEventHistorySize) {
          receiver_event_key_set_.erase(receiver_event_key_queue_.front());
          receiver_event_key_queue_.pop();
        }
      } else {
        messages->erase(tmp);
      }
    }

    RtcpReceiverLogMessage::iterator tmp = i;
    ++i;
    if (messages->empty()) {
      receiver_log->erase(tmp);
    }
  }
  return !receiver_log->empty();
}

void Rtcp::SendRtcpFromRtpReceiver(
    const RtcpCastMessage* cast_message,
    base::TimeDelta target_delay,
    const ReceiverRtcpEventSubscriber::RtcpEventMultiMap* rtcp_events,
    RtpReceiverStatistics* rtp_receiver_statistics) {
  base::TimeTicks now = clock_->NowTicks();
  RtcpReportBlock report_block;
  RtcpReceiverReferenceTimeReport rrtr;

  // Attach our NTP to all RTCP packets; with this information a "smart" sender
  // can make decisions based on how old the RTCP message is.
  ConvertTimeTicksToNtp(now, &rrtr.ntp_seconds, &rrtr.ntp_fraction);
  SaveLastSentNtpTime(now, rrtr.ntp_seconds, rrtr.ntp_fraction);

  if (rtp_receiver_statistics) {
    report_block.remote_ssrc = 0;            // Not needed to set send side.
    report_block.media_ssrc = remote_ssrc_;  // SSRC of the RTP packet sender.
    rtp_receiver_statistics->GetStatistics(
        &report_block.fraction_lost, &report_block.cumulative_lost,
        &report_block.extended_high_sequence_number, &report_block.jitter);

    report_block.last_sr = last_report_truncated_ntp_;
    if (!time_last_report_received_.is_null()) {
      uint32 delay_seconds = 0;
      uint32 delay_fraction = 0;
      base::TimeDelta delta = now - time_last_report_received_;
      ConvertTimeToFractions(delta.InMicroseconds(), &delay_seconds,
                             &delay_fraction);
      report_block.delay_since_last_sr =
          ConvertToNtpDiff(delay_seconds, delay_fraction);
    } else {
      report_block.delay_since_last_sr = 0;
    }
  }
  packet_sender_->SendRtcpPacket(
      local_ssrc_,
      rtcp_builder_.BuildRtcpFromReceiver(
          rtp_receiver_statistics ? &report_block : NULL,
          &rrtr,
          cast_message,
          rtcp_events,
          target_delay));
}

void Rtcp::SendRtcpFromRtpSender(base::TimeTicks current_time,
                                 uint32 current_time_as_rtp_timestamp,
                                 uint32 send_packet_count,
                                 size_t send_octet_count) {
  uint32 current_ntp_seconds = 0;
  uint32 current_ntp_fractions = 0;
  ConvertTimeTicksToNtp(current_time, &current_ntp_seconds,
                        &current_ntp_fractions);
  SaveLastSentNtpTime(current_time, current_ntp_seconds,
                      current_ntp_fractions);

  RtcpSenderInfo sender_info;
  sender_info.ntp_seconds = current_ntp_seconds;
  sender_info.ntp_fraction = current_ntp_fractions;
  sender_info.rtp_timestamp = current_time_as_rtp_timestamp;
  sender_info.send_packet_count = send_packet_count;
  sender_info.send_octet_count = send_octet_count;

  packet_sender_->SendRtcpPacket(
      local_ssrc_,
      rtcp_builder_.BuildRtcpFromSender(sender_info));
}

void Rtcp::OnReceivedNtp(uint32 ntp_seconds, uint32 ntp_fraction) {
  last_report_truncated_ntp_ = ConvertToNtpDiff(ntp_seconds, ntp_fraction);

  const base::TimeTicks now = clock_->NowTicks();
  time_last_report_received_ = now;

  // TODO(miu): This clock offset calculation does not account for packet
  // transit time over the network.  End2EndTest.EvilNetwork confirms that this
  // contributes a very significant source of error here.  Determine whether
  // RTT should be factored-in, and how that changes the rest of the
  // calculation.
  const base::TimeDelta measured_offset =
      now - ConvertNtpToTimeTicks(ntp_seconds, ntp_fraction);
  local_clock_ahead_by_.Update(now, measured_offset);
  if (measured_offset < local_clock_ahead_by_.Current()) {
    // Logically, the minimum offset between the clocks has to be the correct
    // one.  For example, the time it took to transmit the current report may
    // have been lower than usual, and so some of the error introduced by the
    // transmission time can be eliminated.
    local_clock_ahead_by_.Reset(now, measured_offset);
  }
  VLOG(1) << "Local clock is ahead of the remote clock by: "
          << "measured=" << measured_offset.InMicroseconds() << " usec, "
          << "filtered=" << local_clock_ahead_by_.Current().InMicroseconds()
          << " usec.";
}

void Rtcp::OnReceivedLipSyncInfo(uint32 rtp_timestamp, uint32 ntp_seconds,
                                 uint32 ntp_fraction) {
  if (ntp_seconds == 0) {
    NOTREACHED();
    return;
  }
  lip_sync_rtp_timestamp_ = rtp_timestamp;
  lip_sync_ntp_timestamp_ =
      (static_cast<uint64>(ntp_seconds) << 32) | ntp_fraction;
}

bool Rtcp::GetLatestLipSyncTimes(uint32* rtp_timestamp,
                                 base::TimeTicks* reference_time) const {
  if (!lip_sync_ntp_timestamp_)
    return false;

  const base::TimeTicks local_reference_time =
      ConvertNtpToTimeTicks(static_cast<uint32>(lip_sync_ntp_timestamp_ >> 32),
                            static_cast<uint32>(lip_sync_ntp_timestamp_)) +
      local_clock_ahead_by_.Current();

  // Sanity-check: Getting regular lip sync updates?
  DCHECK((clock_->NowTicks() - local_reference_time) <
         base::TimeDelta::FromMinutes(1));

  *rtp_timestamp = lip_sync_rtp_timestamp_;
  *reference_time = local_reference_time;
  return true;
}

void Rtcp::OnReceivedDelaySinceLastReport(uint32 last_report,
                                          uint32 delay_since_last_report) {
  RtcpSendTimeMap::iterator it = last_reports_sent_map_.find(last_report);
  if (it == last_reports_sent_map_.end()) {
    return;  // Feedback on another report.
  }

  const base::TimeDelta sender_delay = clock_->NowTicks() - it->second;
  const base::TimeDelta receiver_delay =
      ConvertFromNtpDiff(delay_since_last_report);
  current_round_trip_time_ = sender_delay - receiver_delay;
  // If the round trip time was computed as less than 1 ms, assume clock
  // imprecision by one or both peers caused a bad value to be calculated.
  // While plenty of networks do easily achieve less than 1 ms round trip time,
  // such a level of precision cannot be measured with our approach; and 1 ms is
  // good enough to represent "under 1 ms" for our use cases.
  current_round_trip_time_ =
      std::max(current_round_trip_time_, base::TimeDelta::FromMilliseconds(1));

  if (!rtt_callback_.is_null())
    rtt_callback_.Run(current_round_trip_time_);
}

void Rtcp::OnReceivedCastFeedback(const RtcpCastMessage& cast_message) {
  if (cast_callback_.is_null())
    return;
  cast_callback_.Run(cast_message);
}

void Rtcp::SaveLastSentNtpTime(const base::TimeTicks& now,
                               uint32 last_ntp_seconds,
                               uint32 last_ntp_fraction) {
  // Make sure |now| is always greater than the last element in
  // |last_reports_sent_queue_|.
  if (!last_reports_sent_queue_.empty())
    DCHECK(now >= last_reports_sent_queue_.back().second);

  uint32 last_report = ConvertToNtpDiff(last_ntp_seconds, last_ntp_fraction);
  last_reports_sent_map_[last_report] = now;
  last_reports_sent_queue_.push(std::make_pair(last_report, now));

  const base::TimeTicks timeout =
      now - TimeDelta::FromMilliseconds(kStatsHistoryWindowMs);

  // Cleanup old statistics older than |timeout|.
  while (!last_reports_sent_queue_.empty()) {
    RtcpSendTimePair oldest_report = last_reports_sent_queue_.front();
    if (oldest_report.second < timeout) {
      last_reports_sent_map_.erase(oldest_report.first);
      last_reports_sent_queue_.pop();
    } else {
      break;
    }
  }
}

void Rtcp::OnReceivedReceiverLog(const RtcpReceiverLogMessage& receiver_log) {
  if (log_callback_.is_null())
    return;
  log_callback_.Run(receiver_log);
}

}  // namespace cast
}  // namespace media