Refactor NACK bitrate allocation

webrtc/modules/rtp_rtcp/source/rtp_sender.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.
  */
 
@@ skipping 15 matching lines @@
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h"
 #include "webrtc/modules/rtp_rtcp/source/time_util.h"
 
 namespace webrtc {
 
 // Max in the RFC 3550 is 255 bytes, we limit it to be modulus 32 for SRTP.
 static const size_t kMaxPaddingLength = 224;
 static const int kSendSideDelayWindowMs = 1000;
 static const uint32_t kAbsSendTimeFraction = 18;
+static const int kBitrateStatisticsWindowMs = 1000;
 
 namespace {
 
 const size_t kRtpHeaderLength = 12;
 const uint16_t kMaxInitRtpSeqNumber = 32767;  // 2^15 -1.
 
 const char* FrameTypeToString(FrameType frame_type) {
   switch (frame_type) {
     case kEmptyFrame:
       return "empty";
@@ skipping 10 matching lines @@
 uint32_t ConvertMsTo24Bits(int64_t time_ms) {
   uint32_t time_24_bits =
       static_cast<uint32_t>(
           ((static_cast<uint64_t>(time_ms) << kAbsSendTimeFraction) + 500) /
           1000) &
       0x00FFFFFF;
   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,
     Clock* clock,
     Transport* transport,
     RtpPacketSender* paced_sender,
     TransportSequenceNumberAllocator* sequence_number_allocator,
     TransportFeedbackObserver* transport_feedback_observer,
     BitrateStatisticsObserver* bitrate_callback,
     FrameCountObserver* frame_count_observer,
     SendSideDelayObserver* send_side_delay_observer,
     RtcEventLog* event_log,
-    SendPacketObserver* send_packet_observer)
+    SendPacketObserver* send_packet_observer,
+    NackRateLimiter* nack_rate_limiter)
     : 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)),
       paced_sender_(paced_sender),
       transport_sequence_number_allocator_(sequence_number_allocator),
       transport_feedback_observer_(transport_feedback_observer),
       last_capture_time_ms_sent_(0),
       transport_(transport),
       sending_media_(true),                      // Default to sending media.
       max_payload_length_(IP_PACKET_SIZE - 28),  // Default is IP-v4/UDP.
       payload_type_(-1),
       payload_type_map_(),
       rtp_header_extension_map_(),
       transmission_time_offset_(0),
       absolute_send_time_(0),
       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_(NULL),
+      rtp_stats_callback_(nullptr),
+      total_bitrate_sent_(kBitrateStatisticsWindowMs,
+                          RateStatistics::kBpsScale),
+      nack_bitrate_sent_(kBitrateStatisticsWindowMs, RateStatistics::kBpsScale),
       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),
       ssrc_db_(SSRCDatabase::GetSSRCDatabase()),
       remote_ssrc_(0),
       sequence_number_forced_(false),
       ssrc_forced_(false),
       timestamp_(0),
       capture_time_ms_(0),
       last_timestamp_time_ms_(0),
       media_has_been_sent_(false),
       last_packet_marker_bit_(false),
       csrcs_(),
       rtx_(kRtxOff),
-      target_bitrate_(0) {
-  memset(nack_byte_count_times_, 0, sizeof(nack_byte_count_times_));
-  memset(nack_byte_count_, 0, sizeof(nack_byte_count_));
+      target_bitrate_(0),
+      nack_rate_limiter_(nack_rate_limiter) {
   // 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.
   srand(static_cast<uint32_t>(clock_->TimeInMilliseconds()));
   ssrc_ = ssrc_db_->CreateSSRC();
   RTC_DCHECK(ssrc_ != 0);
   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);
 }
 
 RTPSender::~RTPSender() {
   // TODO(tommi): Use a thread checker to ensure the object is created and
   // deleted on the same thread.  At the moment this isn't possible due to
   // voe::ChannelOwner in voice engine.  To reproduce, run:
   // voe_auto_test --automated --gtest_filter=*MixManyChannelsForStressOpus
@@ skipping 20 matching lines @@
   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 {
-  return (uint16_t)(total_bitrate_sent_.BitrateNow() / 1000);
+  rtc::CritScope cs(&statistics_crit_);
+  return static_cast<uint16_t>(
+      total_bitrate_sent_.Rate(clock_->TimeInMilliseconds()).value_or(0) /
+      1000);
 }
 
 uint32_t RTPSender::VideoBitrateSent() const {
   if (video_) {
     return video_->VideoBitrateSent();
   }
   return 0;
 }
 
 uint32_t RTPSender::FecOverheadRate() const {
   if (video_) {
     return video_->FecOverheadRate();
   }
   return 0;
 }
 
 uint32_t RTPSender::NackOverheadRate() const {
-  return nack_bitrate_.BitrateLast();
+  rtc::CritScope cs(&statistics_crit_);
+  return nack_bitrate_sent_.Rate(clock_->TimeInMilliseconds()).value_or(0);
 }
 
 int32_t RTPSender::SetTransmissionTimeOffset(int32_t transmission_time_offset) {
   if (transmission_time_offset > (0x800000 - 1) ||
       transmission_time_offset < -(0x800000 - 1)) {  // Word24.
     return -1;
   }
   rtc::CritScope lock(&send_critsect_);
   transmission_time_offset_ = transmission_time_offset;
   return 0;
@@ skipping 495 matching lines @@
   uint8_t data_buffer[IP_PACKET_SIZE];
   int64_t capture_time_ms;
 
   if (!packet_history_.GetPacketAndSetSendTime(packet_id, min_resend_time, true,
                                                data_buffer, &length,
                                                &capture_time_ms)) {
     // Packet not found.
     return 0;
   }
 
+  // Check if we're overusing retransmission bitrate.
+  // TODO(sprang): Add histograms for nack success or failure reasons.
+  if (nack_rate_limiter_ && !nack_rate_limiter_->TryUseRate(length))
+    return -1;
+
   if (paced_sender_) {
     RtpUtility::RtpHeaderParser rtp_parser(data_buffer, length);
     RTPHeader header;
     if (!rtp_parser.Parse(&header)) {
       assert(false);
       return -1;
     }
     // Convert from TickTime to Clock since capture_time_ms is based on
     // TickTime.
     int64_t corrected_capture_tims_ms = capture_time_ms + clock_delta_ms_;
@@ skipping 50 matching lines @@
     return -1;
   video_->SetSelectiveRetransmissions(settings);
   return 0;
 }
 
 void RTPSender::OnReceivedNACK(const std::list<uint16_t>& nack_sequence_numbers,
                                int64_t avg_rtt) {
   TRACE_EVENT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
                "RTPSender::OnReceivedNACK", "num_seqnum",
                nack_sequence_numbers.size(), "avg_rtt", avg_rtt);
-  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) {
+  for (uint16_t seq_no : nack_sequence_numbers) {
+    const int32_t bytes_sent = ReSendPacket(seq_no, 5 + avg_rtt);
+    if (bytes_sent < 0) {
+      // Failed to send one Sequence number. Give up the rest in this nack.
+      LOG(LS_WARNING) << "Failed resending RTP packet " << seq_no
+                      << ", Discard rest of packets";
+      break;
+    }
+    if (bytes_sent == 0) {

[danilchap, 2016/06/28 14:26:18]

this block not needed now.

[sprang_webrtc, 2016/07/04 09:33:04]

Done.

       // 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 " << *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.
 bool RTPSender::TimeToSendPacket(uint16_t sequence_number,
                                  int64_t capture_time_ms,
                                  bool retransmission,
                                  int probe_cluster_id) {
   size_t length = IP_PACKET_SIZE;
   uint8_t data_buffer[IP_PACKET_SIZE];
   int64_t stored_time_ms;
 
   if (!packet_history_.GetPacketAndSetSendTime(sequence_number,
@@ skipping 74 matching lines @@
 }
 
 void RTPSender::UpdateRtpStats(const uint8_t* buffer,
                                size_t packet_length,
                                const RTPHeader& header,
                                bool is_rtx,
                                bool is_retransmit) {
   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) {
     counters = &rtx_rtp_stats_;
   } else {
     counters = &rtp_stats_;
   }
 
-  total_bitrate_sent_.Update(packet_length);
+  total_bitrate_sent_.Update(packet_length, now_ms);
 
-  if (counters->first_packet_time_ms == -1) {
+  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,
                             const RTPHeader& header) const {
   if (!video_) {
     return false;
   }
   bool fec_enabled;
   uint8_t pt_red;
   uint8_t pt_fec;
@@ skipping 127 matching lines @@
 void RTPSender::UpdateOnSendPacket(int packet_id,
                                    int64_t capture_time_ms,
                                    uint32_t ssrc) {
   if (!send_packet_observer_ || capture_time_ms <= 0 || packet_id == -1)
     return;
 
   send_packet_observer_->OnSendPacket(packet_id, capture_time_ms, ssrc);
 }
 
 void RTPSender::ProcessBitrate() {
-  rtc::CritScope lock(&send_critsect_);
-  total_bitrate_sent_.Process();
-  nack_bitrate_.Process();
-  if (audio_configured_) {
+  if (!bitrate_callback_)
     return;
+  int64_t now_ms = clock_->TimeInMilliseconds();
+  uint32_t ssrc;
+  {
+    rtc::CritScope lock(&send_critsect_);
+    ssrc = ssrc_;
   }
-  video_->ProcessBitrate();
+
+  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);
 }
 
 size_t RTPSender::RtpHeaderLength() const {
   rtc::CritScope lock(&send_critsect_);
   size_t rtp_header_length = kRtpHeaderLength;
   rtp_header_length += sizeof(uint32_t) * csrcs_.size();
   rtp_header_length += RtpHeaderExtensionLength();
   return rtp_header_length;
 }
 
@@ skipping 539 matching lines @@
 
     // Will be ignored if it's already configured via API.
     SetStartTimestamp(RTPtime, false);
   } else {
     rtc::CritScope lock(&send_critsect_);
     if (!ssrc_forced_) {
       // Generate a new SSRC.
       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_) {
       // Generate a new sequence number.
       sequence_number_ = random_.Rand(1, kMaxInitRtpSeqNumber);
     }
   }
 }
 
 void RTPSender::SetSendingMediaStatus(bool enabled) {
@@ skipping 30 matching lines @@
 
 uint32_t RTPSender::GenerateNewSSRC() {
   // If configured via API, return 0.
   rtc::CritScope lock(&send_critsect_);
 
   if (ssrc_forced_) {
     return 0;
   }
   ssrc_ = ssrc_db_->CreateSSRC();
   RTC_DCHECK(ssrc_ != 0);
-  bitrates_.set_ssrc(ssrc_);
   return ssrc_;
 }
 
 void RTPSender::SetSSRC(uint32_t ssrc) {
   // This is configured via the API.
   rtc::CritScope lock(&send_critsect_);
 
   if (ssrc_ == ssrc && ssrc_forced_) {
     return;  // Since it's same ssrc, don't reset anything.
   }
   ssrc_forced_ = true;
   ssrc_db_->ReturnSSRC(ssrc_);
   ssrc_db_->RegisterSSRC(ssrc);
   ssrc_ = ssrc;
-  bitrates_.set_ssrc(ssrc_);
   if (!sequence_number_forced_) {
     sequence_number_ = random_.Rand(1, kMaxInitRtpSeqNumber);
   }
 }
 
 uint32_t RTPSender::SSRC() const {
   rtc::CritScope lock(&send_critsect_);
   return ssrc_;
 }
 
@@ skipping 128 matching lines @@
   rtc::CritScope cs(&statistics_crit_);
   rtp_stats_callback_ = callback;
 }
 
 StreamDataCountersCallback* RTPSender::GetRtpStatisticsCallback() const {
   rtc::CritScope cs(&statistics_crit_);
   return rtp_stats_callback_;
 }
 
 uint32_t RTPSender::BitrateSent() const {
-  return total_bitrate_sent_.BitrateLast();
+  rtc::CritScope cs(&statistics_crit_);
+  return total_bitrate_sent_.Rate(clock_->TimeInMilliseconds()).value_or(0);
 }
 
 void RTPSender::SetRtpState(const RtpState& rtp_state) {
   rtc::CritScope lock(&send_critsect_);
   sequence_number_ = rtp_state.sequence_number;
   sequence_number_forced_ = true;
   timestamp_ = rtp_state.timestamp;
   capture_time_ms_ = rtp_state.capture_time_ms;
   last_timestamp_time_ms_ = rtp_state.last_timestamp_time_ms;
   media_has_been_sent_ = rtp_state.media_has_been_sent;
@@ skipping 22 matching lines @@
   rtc::CritScope lock(&send_critsect_);
 
   RtpState state;
   state.sequence_number = sequence_number_rtx_;
   state.start_timestamp = start_timestamp_;
 
   return state;
 }
 
 }  // namespace webrtc