cast: Split Rtcp into two for sender and receiver

media/cast/net/rtcp/sender_rtcp_session.cc

+// Copyright 2015 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 <algorithm>
+#include <limits>
+#include <utility>
+
+#include "base/big_endian.h"
+#include "base/time/time.h"
+#include "media/cast/constants.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"
+#include "media/cast/net/rtcp/sender_rtcp_session.h"
+
+namespace media {
+namespace cast {
+
+namespace {
+
+enum {
+  kStatsHistoryWindowMs = 10000,  // 10 seconds.
+
+  // Reject packets that are 0.5 seconds older than
+  // the newest packet we've seen so far. This protects internal
+  // states from crazy routers. (Based on RRTR)
+  kOutOfOrderMaxAgeMs = 500,
+};
+
+// Create a NTP diff from seconds and fractions of seconds; delay_fraction is
+// fractions of a second where 0x80000000 is half a second.
+uint32_t ConvertToNtpDiff(uint32_t delay_seconds, uint32_t delay_fraction) {
+  return ((delay_seconds & 0x0000FFFF) << 16) +
+         ((delay_fraction & 0xFFFF0000) >> 16);
+}
+
+// Parse a NTP diff value into a base::TimeDelta.
+base::TimeDelta ConvertFromNtpDiff(uint32_t ntp_delay) {
+  int64_t delay_us =
+      (ntp_delay & 0x0000ffff) * base::Time::kMicrosecondsPerSecond;
+  delay_us >>= 16;
+  delay_us +=
+      ((ntp_delay & 0xffff0000) >> 16) * base::Time::kMicrosecondsPerSecond;
+  return base::TimeDelta::FromMicroseconds(delay_us);
+}
+
+// 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_t:
+//   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_t:
+//   bits 0-63: event TimeTicks internal value.
+std::pair<uint64_t, uint64_t> GetReceiverEventKey(
+    RtpTimeTicks frame_rtp_timestamp,
+    const base::TimeTicks& event_timestamp,
+    uint8_t event_type,
+    uint16_t packet_id_or_zero) {
+  uint64_t value1 = event_type;
+  value1 <<= 16;
+  value1 |= packet_id_or_zero;
+  value1 <<= 32;
+  value1 |= frame_rtp_timestamp.lower_32_bits();
+  return std::make_pair(
+      value1, static_cast<uint64_t>(event_timestamp.ToInternalValue()));
+}
+
+}  // namespace
+
+SenderRtcpSession::SenderRtcpSession(
+    const RtcpCastMessageCallback& cast_callback,
+    const RtcpRttCallback& rtt_callback,
+    const RtcpLogMessageCallback& log_callback,
+    base::TickClock* clock,
+    PacedPacketSender* packet_sender,
+    uint32_t local_ssrc,
+    uint32_t remote_ssrc)
+    : clock_(clock),
+      packet_sender_(packet_sender),
+      local_ssrc_(local_ssrc),
+      remote_ssrc_(remote_ssrc),
+      cast_callback_(cast_callback),
+      rtt_callback_(rtt_callback),
+      log_callback_(log_callback),
+      largest_seen_timestamp_(base::TimeTicks::FromInternalValue(
+          std::numeric_limits<int64_t>::min())),
+      parser_(local_ssrc, remote_ssrc),
+      ack_frame_id_wrap_helper_(kFirstFrameId - 1) {}
+
+SenderRtcpSession::~SenderRtcpSession() {}
+
+bool SenderRtcpSession::IncomingRtcpPacket(const uint8_t* 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_t ssrc_of_sender = GetSsrcOfSender(data, length);
+  if (ssrc_of_sender != remote_ssrc_) {
+    return false;
+  }
+
+  // Parse this packet.
+  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;
+      }
+    }
+    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()) {
+      VLOG(1) << "1 sender_rtcp_session "
+              << parser_.mutable_cast_message()->ack_frame_id;
+      parser_.mutable_cast_message()->ack_frame_id =
+          ack_frame_id_wrap_helper_.MapTo32bitsFrameId(
+              parser_.mutable_cast_message()->ack_frame_id);
+      VLOG(1) << "2 sender_rtcp_session "
+              << parser_.mutable_cast_message()->ack_frame_id;
+      OnReceivedCastFeedback(parser_.cast_message());
+    }
+  }
+  return true;
+}
+
+void SenderRtcpSession::OnReceivedDelaySinceLastReport(
+    uint32_t last_report,
+    uint32_t 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 SenderRtcpSession::SaveLastSentNtpTime(const base::TimeTicks& now,
+                                            uint32_t last_ntp_seconds,
+                                            uint32_t 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_t 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 - base::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;
+    }
+  }
+}
+
+bool SenderRtcpSession::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 SenderRtcpSession::SendRtcpReport(
+    base::TimeTicks current_time,
+    RtpTimeTicks current_time_as_rtp_timestamp,
+    uint32_t send_packet_count,
+    size_t send_octet_count) {
+  uint32_t current_ntp_seconds = 0;
+  uint32_t 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;
+
+  RtcpBuilder rtcp_builder(local_ssrc_);
+  packet_sender_->SendRtcpPacket(local_ssrc_,
+                                 rtcp_builder.BuildRtcpFromSender(sender_info));
+}
+
+void SenderRtcpSession::OnReceivedCastFeedback(
+    const RtcpCastMessage& cast_message) {
+  if (cast_callback_.is_null())
+    return;
+  cast_callback_.Run(cast_message);
+}
+
+void SenderRtcpSession::OnReceivedReceiverLog(
+    const RtcpReceiverLogMessage& receiver_log) {
+  if (log_callback_.is_null())
+    return;
+  log_callback_.Run(receiver_log);
+}
+
+}  // namespace cast
+}  // namespace media