Add TransportFeedback adapter, adapting remote feedback to bwe estiamtor

webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.cc

Index: webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.cc
diff --git a/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.cc b/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.cc
new file mode 100644
index 0000000000000000000000000000000000000000..c6e34f212dae527f8e304bdd0a11f03429c28bd6
--- /dev/null
+++ b/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.cc
@@ -0,0 +1,126 @@
+/*
+ *  Copyright (c) 2015 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 <limits>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h"
+#include "webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
+#include "webrtc/modules/utility/interface/process_thread.h"
+
+namespace webrtc {
+
+const int64_t kNoTimestamp = -1;
+const int64_t kSendTimeHistoryWindowMs = 10000;
+const int64_t kBaseTimestampScaleFactor =
+    rtcp::TransportFeedback::kDeltaScaleFactor * (1 << 8);
+const int64_t kBaseTimestampRangeSizeUs = kBaseTimestampScaleFactor * (1 << 24);
+
+TransportFeedbackAdapter::TransportFeedbackAdapter(
+    RtcpBandwidthObserver* bandwidth_observer,
+    Clock* clock,
+    ProcessThread* process_thread)
+    : send_time_history_(kSendTimeHistoryWindowMs),
+      rtcp_bandwidth_observer_(bandwidth_observer),
+      process_thread_(process_thread),
+      clock_(clock),
+      current_offset_ms_(kNoTimestamp),
+      last_timestamp_us_(kNoTimestamp) {}
+
+TransportFeedbackAdapter::~TransportFeedbackAdapter() {
+  if (bitrate_estimator_.get())
+    process_thread_->DeRegisterModule(bitrate_estimator_.get());
+}
+
+void TransportFeedbackAdapter::SetBitrateEstimator(
+    RemoteBitrateEstimator* rbe) {
+  if (bitrate_estimator_.get() != rbe) {
+    bitrate_estimator_.reset(rbe);
+    process_thread_->RegisterModule(rbe);
+  }
+}
+
+void TransportFeedbackAdapter::OnPacketSent(const PacketInfo& info) {
+  rtc::CritScope cs(&lock_);
+  send_time_history_.AddAndRemoveOld(info);
+}
+
+void TransportFeedbackAdapter::OnTransportFeedback(
+    const rtcp::TransportFeedback& feedback) {
+  int64_t timestamp_us = feedback.GetBaseTimeUs();
+  // Add timestamp deltas to a local time base selected on first packet arrival.
+  // This won't be the true time base, but makes it easier to manually inspect
+  // time stamps.
+  if (last_timestamp_us_ == kNoTimestamp) {
+    current_offset_ms_ = clock_->TimeInMilliseconds();
+  } else {
+    int64_t delta = timestamp_us - last_timestamp_us_;
+
+    // Detect and compensate for wrap-arounds in base time.
+    if (std::abs(delta - kBaseTimestampRangeSizeUs) < std::abs(delta)) {
+      delta -= kBaseTimestampRangeSizeUs;  // Wrap backwards.
+    } else if (std::abs(delta + kBaseTimestampRangeSizeUs) < std::abs(delta)) {
+      delta += kBaseTimestampRangeSizeUs;  // Wrap forwards.
+    }
+
+    current_offset_ms_ += delta / 1000;
+  }
+  last_timestamp_us_ = timestamp_us;
+
+  uint16_t sequence_number = feedback.GetBaseSequence();
+  std::vector<int64_t> delta_vec = feedback.GetReceiveDeltasUs();
+  auto delta_it = delta_vec.begin();
+  std::vector<PacketInfo> packet_feedback_vector;
+  packet_feedback_vector.reserve(delta_vec.size());
+
+  {
+    rtc::CritScope cs(&lock_);
+    size_t failed_lookups = 0;
+    int64_t offset_us = 0;
+    for (auto symbol : feedback.GetStatusVector()) {
+      if (symbol != rtcp::TransportFeedback::StatusSymbol::kNotReceived) {
+        DCHECK(delta_it != delta_vec.end());
+        offset_us += *(delta_it++);
+        int64_t timestamp_ms = current_offset_ms_ + (offset_us / 1000);
+        PacketInfo info = {timestamp_ms, 0, sequence_number, 0, false};
+        if (send_time_history_.GetInfo(&info, true)) {
+          packet_feedback_vector.push_back(info);
+        } else {
+          ++failed_lookups;
+        }
+      }
+      ++sequence_number;
+    }
+    DCHECK(delta_it == delta_vec.end());
+    if (failed_lookups > 0) {
+      LOG(LS_WARNING) << "Failed to lookup send time for " << failed_lookups

[kjellander_webrtc, 2016/09/29 05:15:51]

This warning can spam the logs a lot during a call. Can you implement some of
kind of throttling for it?

+                      << " packet" << (failed_lookups > 1 ? "s" : "")
+                      << ". Send time history too small?";
+    }
+  }
+  DCHECK(bitrate_estimator_.get() != nullptr);
+  bitrate_estimator_->IncomingPacketFeedbackVector(packet_feedback_vector);
+}
+
+void TransportFeedbackAdapter::OnReceiveBitrateChanged(
+    const std::vector<unsigned int>& ssrcs,
+    unsigned int bitrate) {
+  rtcp_bandwidth_observer_->OnReceivedEstimatedBitrate(bitrate);
+}
+
+void TransportFeedbackAdapter::OnRttUpdate(int64_t avg_rtt_ms,
+                                           int64_t max_rtt_ms) {
+  DCHECK(bitrate_estimator_.get() != nullptr);
+  bitrate_estimator_->OnRttUpdate(avg_rtt_ms, max_rtt_ms);
+}
+
+}  // namespace webrtc