Revert of Refactor NACK bitrate allocation

webrtc/modules/rtp_rtcp/source/rtp_sender.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_sender.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index 25016e053f2b43a948f803846fda364423d31dbe..4ee2524abc3153c8103fa9d0377e1e056d13b6f9 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -16,7 +16,6 @@
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
-#include "webrtc/base/rate_limiter.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/call.h"
@@ -34,7 +33,6 @@
 static const size_t kMaxPaddingLength = 224;
 static const int kSendSideDelayWindowMs = 1000;
 static const uint32_t kAbsSendTimeFraction = 18;
-static const int kBitrateStatisticsWindowMs = 1000;
 
 namespace {
 
@@ -64,6 +62,47 @@
   return time_24_bits;
 }
 }  // namespace
+
+RTPSender::BitrateAggregator::BitrateAggregator(
+    BitrateStatisticsObserver* bitrate_callback)
+    : callback_(bitrate_callback),
+      total_bitrate_observer_(*this),
+      retransmit_bitrate_observer_(*this),
+      ssrc_(0) {}
+
+void RTPSender::BitrateAggregator::OnStatsUpdated() const {
+  if (callback_) {
+    callback_->Notify(total_bitrate_observer_.statistics(),
+                      retransmit_bitrate_observer_.statistics(), ssrc_);
+  }
+}
+
+Bitrate::Observer* RTPSender::BitrateAggregator::total_bitrate_observer() {
+  return &total_bitrate_observer_;
+}
+Bitrate::Observer* RTPSender::BitrateAggregator::retransmit_bitrate_observer() {
+  return &retransmit_bitrate_observer_;
+}
+
+void RTPSender::BitrateAggregator::set_ssrc(uint32_t ssrc) {
+  ssrc_ = ssrc;
+}
+
+RTPSender::BitrateAggregator::BitrateObserver::BitrateObserver(
+    const BitrateAggregator& aggregator)
+    : aggregator_(aggregator) {}
+
+// Implements Bitrate::Observer.
+void RTPSender::BitrateAggregator::BitrateObserver::BitrateUpdated(
+    const BitrateStatistics& stats) {
+  statistics_ = stats;
+  aggregator_.OnStatsUpdated();
+}
+
+const BitrateStatistics&
+RTPSender::BitrateAggregator::BitrateObserver::statistics() const {
+  return statistics_;
+}
 
 RTPSender::RTPSender(
     bool audio,
@@ -76,12 +115,13 @@
     FrameCountObserver* frame_count_observer,
     SendSideDelayObserver* send_side_delay_observer,
     RtcEventLog* event_log,
-    SendPacketObserver* send_packet_observer,
-    RateLimiter* retransmission_rate_limiter)
+    SendPacketObserver* send_packet_observer)
     : clock_(clock),
       // TODO(holmer): Remove this conversion?
       clock_delta_ms_(clock_->TimeInMilliseconds() - rtc::TimeMillis()),
       random_(clock_->TimeInMicroseconds()),
+      bitrates_(bitrate_callback),
+      total_bitrate_sent_(clock, bitrates_.total_bitrate_observer()),
       audio_configured_(audio),
       audio_(audio ? new RTPSenderAudio(clock, this) : nullptr),
       video_(audio ? nullptr : new RTPSenderVideo(clock, this)),
@@ -100,18 +140,18 @@
       rotation_(kVideoRotation_0),
       video_rotation_active_(false),
       transport_sequence_number_(0),
+      // NACK.
+      nack_byte_count_times_(),
+      nack_byte_count_(),
+      nack_bitrate_(clock, bitrates_.retransmit_bitrate_observer()),
       playout_delay_active_(false),
       packet_history_(clock),
       // Statistics
-      rtp_stats_callback_(nullptr),
-      total_bitrate_sent_(kBitrateStatisticsWindowMs,
-                          RateStatistics::kBpsScale),
-      nack_bitrate_sent_(kBitrateStatisticsWindowMs, RateStatistics::kBpsScale),
+      rtp_stats_callback_(NULL),
       frame_count_observer_(frame_count_observer),
       send_side_delay_observer_(send_side_delay_observer),
       event_log_(event_log),
       send_packet_observer_(send_packet_observer),
-      bitrate_callback_(bitrate_callback),
       // RTP variables
       start_timestamp_forced_(false),
       start_timestamp_(0),
@@ -126,7 +166,9 @@
       last_packet_marker_bit_(false),
       csrcs_(),
       rtx_(kRtxOff),
-      retransmission_rate_limiter_(retransmission_rate_limiter) {
+      target_bitrate_(0) {
+  memset(nack_byte_count_times_, 0, sizeof(nack_byte_count_times_));
+  memset(nack_byte_count_, 0, sizeof(nack_byte_count_));
   // We need to seed the random generator for BuildPaddingPacket() below.
   // TODO(holmer,tommi): Note that TimeInMilliseconds might return 0 on Mac
   // early on in the process.
@@ -136,6 +178,7 @@
   ssrc_rtx_ = ssrc_db_->CreateSSRC();
   RTC_DCHECK(ssrc_rtx_ != 0);
 
+  bitrates_.set_ssrc(ssrc_);
   // Random start, 16 bits. Can't be 0.
   sequence_number_rtx_ = random_.Rand(1, kMaxInitRtpSeqNumber);
   sequence_number_ = random_.Rand(1, kMaxInitRtpSeqNumber);
@@ -165,11 +208,18 @@
   }
 }
 
+void RTPSender::SetTargetBitrate(uint32_t bitrate) {
+  rtc::CritScope cs(&target_bitrate_critsect_);
+  target_bitrate_ = bitrate;
+}
+
+uint32_t RTPSender::GetTargetBitrate() {
+  rtc::CritScope cs(&target_bitrate_critsect_);
+  return target_bitrate_;
+}
+
 uint16_t RTPSender::ActualSendBitrateKbit() const {
-  rtc::CritScope cs(&statistics_crit_);
-  return static_cast<uint16_t>(
-      total_bitrate_sent_.Rate(clock_->TimeInMilliseconds()).value_or(0) /
-      1000);
+  return (uint16_t)(total_bitrate_sent_.BitrateNow() / 1000);
 }
 
 uint32_t RTPSender::VideoBitrateSent() const {
@@ -187,8 +237,7 @@
 }
 
 uint32_t RTPSender::NackOverheadRate() const {
-  rtc::CritScope cs(&statistics_crit_);
-  return nack_bitrate_sent_.Rate(clock_->TimeInMilliseconds()).value_or(0);
+  return nack_bitrate_.BitrateLast();
 }
 
 int32_t RTPSender::SetTransmissionTimeOffset(int32_t transmission_time_offset) {
@@ -705,12 +754,6 @@
     return 0;
   }
 
-  // Check if we're overusing retransmission bitrate.
-  // TODO(sprang): Add histograms for nack success or failure reasons.
-  RTC_DCHECK(retransmission_rate_limiter_);
-  if (!retransmission_rate_limiter_->TryUseRate(length))
-    return -1;
-
   if (paced_sender_) {
     RtpUtility::RtpHeaderParser rtp_parser(data_buffer, length);
     RTPHeader header;
@@ -781,20 +824,95 @@
   TRACE_EVENT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
                "RTPSender::OnReceivedNACK", "num_seqnum",
                nack_sequence_numbers.size(), "avg_rtt", avg_rtt);
-  for (uint16_t seq_no : nack_sequence_numbers) {
-    const int32_t bytes_sent = ReSendPacket(seq_no, 5 + avg_rtt);
-    if (bytes_sent < 0) {
+  const int64_t now = clock_->TimeInMilliseconds();
+  uint32_t bytes_re_sent = 0;
+  uint32_t target_bitrate = GetTargetBitrate();
+
+  // Enough bandwidth to send NACK?
+  if (!ProcessNACKBitRate(now)) {
+    LOG(LS_INFO) << "NACK bitrate reached. Skip sending NACK response. Target "
+                 << target_bitrate;
+    return;
+  }
+
+  for (std::list<uint16_t>::const_iterator it = nack_sequence_numbers.begin();
+      it != nack_sequence_numbers.end(); ++it) {
+    const int32_t bytes_sent = ReSendPacket(*it, 5 + avg_rtt);
+    if (bytes_sent > 0) {
+      bytes_re_sent += bytes_sent;
+    } else if (bytes_sent == 0) {
+      // The packet has previously been resent.
+      // Try resending next packet in the list.
+      continue;
+    } else {
       // Failed to send one Sequence number. Give up the rest in this nack.
-      LOG(LS_WARNING) << "Failed resending RTP packet " << seq_no
+      LOG(LS_WARNING) << "Failed resending RTP packet " << *it
                       << ", Discard rest of packets";
       break;
     }
+    // Delay bandwidth estimate (RTT * BW).
+    if (target_bitrate != 0 && avg_rtt) {
+      // kbits/s * ms = bits => bits/8 = bytes
+      size_t target_bytes =
+          (static_cast<size_t>(target_bitrate / 1000) * avg_rtt) >> 3;
+      if (bytes_re_sent > target_bytes) {
+        break;  // Ignore the rest of the packets in the list.
+      }
+    }
+  }
+  if (bytes_re_sent > 0) {
+    UpdateNACKBitRate(bytes_re_sent, now);
   }
 }
 
 void RTPSender::OnReceivedRtcpReportBlocks(
     const ReportBlockList& report_blocks) {
   playout_delay_oracle_.OnReceivedRtcpReportBlocks(report_blocks);
+}
+
+bool RTPSender::ProcessNACKBitRate(uint32_t now) {
+  uint32_t num = 0;
+  size_t byte_count = 0;
+  const uint32_t kAvgIntervalMs = 1000;
+  uint32_t target_bitrate = GetTargetBitrate();
+
+  rtc::CritScope lock(&send_critsect_);
+
+  if (target_bitrate == 0) {
+    return true;
+  }
+  for (num = 0; num < NACK_BYTECOUNT_SIZE; ++num) {
+    if ((now - nack_byte_count_times_[num]) > kAvgIntervalMs) {
+      // Don't use data older than 1sec.
+      break;
+    } else {
+      byte_count += nack_byte_count_[num];
+    }
+  }
+  uint32_t time_interval = kAvgIntervalMs;
+  if (num == NACK_BYTECOUNT_SIZE) {
+    // More than NACK_BYTECOUNT_SIZE nack messages has been received
+    // during the last msg_interval.
+    if (nack_byte_count_times_[num - 1] <= now) {
+      time_interval = now - nack_byte_count_times_[num - 1];
+    }
+  }
+  return (byte_count * 8) < (target_bitrate / 1000 * time_interval);
+}
+
+void RTPSender::UpdateNACKBitRate(uint32_t bytes, int64_t now) {
+  rtc::CritScope lock(&send_critsect_);
+  if (bytes == 0)
+    return;
+  nack_bitrate_.Update(bytes);
+  // Save bitrate statistics.
+  // Shift all but first time.
+  for (int i = NACK_BYTECOUNT_SIZE - 2; i >= 0; i--) {
+    nack_byte_count_[i + 1] = nack_byte_count_[i];
+    nack_byte_count_times_[i + 1] = nack_byte_count_times_[i];
+  }
+  nack_byte_count_[0] = bytes;
+  nack_byte_count_times_[0] = now;
 }
 
 // Called from pacer when we can send the packet.
@@ -891,7 +1009,6 @@
   StreamDataCounters* counters;
   // Get ssrc before taking statistics_crit_ to avoid possible deadlock.
   uint32_t ssrc = is_rtx ? RtxSsrc() : SSRC();
-  int64_t now_ms = clock_->TimeInMilliseconds();
 
   rtc::CritScope lock(&statistics_crit_);
   if (is_rtx) {
@@ -900,23 +1017,22 @@
     counters = &rtp_stats_;
   }
 
-  total_bitrate_sent_.Update(packet_length, now_ms);
-
-  if (counters->first_packet_time_ms == -1)
+  total_bitrate_sent_.Update(packet_length);
+
+  if (counters->first_packet_time_ms == -1) {
     counters->first_packet_time_ms = clock_->TimeInMilliseconds();
-
-  if (IsFecPacket(buffer, header))
+  }
+  if (IsFecPacket(buffer, header)) {
     counters->fec.AddPacket(packet_length, header);
-
+  }
   if (is_retransmit) {
     counters->retransmitted.AddPacket(packet_length, header);
-    nack_bitrate_sent_.Update(packet_length, now_ms);
-  }
-
+  }
   counters->transmitted.AddPacket(packet_length, header);
 
-  if (rtp_stats_callback_)
+  if (rtp_stats_callback_) {
     rtp_stats_callback_->DataCountersUpdated(*counters, ssrc);
+  }
 }
 
 bool RTPSender::IsFecPacket(const uint8_t* buffer,
@@ -1064,18 +1180,13 @@
 }
 
 void RTPSender::ProcessBitrate() {
-  if (!bitrate_callback_)
+  rtc::CritScope lock(&send_critsect_);
+  total_bitrate_sent_.Process();
+  nack_bitrate_.Process();
+  if (audio_configured_) {
     return;
-  int64_t now_ms = clock_->TimeInMilliseconds();
-  uint32_t ssrc;
-  {
-    rtc::CritScope lock(&send_critsect_);
-    ssrc = ssrc_;
-  }
-
-  rtc::CritScope lock(&statistics_crit_);
-  bitrate_callback_->Notify(total_bitrate_sent_.Rate(now_ms).value_or(0),
-                            nack_bitrate_sent_.Rate(now_ms).value_or(0), ssrc);
+  }
+  video_->ProcessBitrate();
 }
 
 size_t RTPSender::RtpHeaderLength() const {
@@ -1635,6 +1746,7 @@
       ssrc_db_->ReturnSSRC(ssrc_);
       ssrc_ = ssrc_db_->CreateSSRC();
       RTC_DCHECK(ssrc_ != 0);
+      bitrates_.set_ssrc(ssrc_);
     }
     // Don't initialize seq number if SSRC passed externally.
     if (!sequence_number_forced_ && !ssrc_forced_) {
@@ -1685,6 +1797,7 @@
   }
   ssrc_ = ssrc_db_->CreateSSRC();
   RTC_DCHECK(ssrc_ != 0);
+  bitrates_.set_ssrc(ssrc_);
   return ssrc_;
 }
 
@@ -1699,6 +1812,7 @@
   ssrc_db_->ReturnSSRC(ssrc_);
   ssrc_db_->RegisterSSRC(ssrc);
   ssrc_ = ssrc;
+  bitrates_.set_ssrc(ssrc_);
   if (!sequence_number_forced_) {
     sequence_number_ = random_.Rand(1, kMaxInitRtpSeqNumber);
   }
@@ -1847,8 +1961,7 @@
 }
 
 uint32_t RTPSender::BitrateSent() const {
-  rtc::CritScope cs(&statistics_crit_);
-  return total_bitrate_sent_.Rate(clock_->TimeInMilliseconds()).value_or(0);
+  return total_bitrate_sent_.BitrateLast();
 }
 
 void RTPSender::SetRtpState(const RtpState& rtp_state) {