1 files changed, 197 insertions, 0 deletions
diff --git a/dom/media/webrtc/libwebrtcglue/RtpSourceObserver.cpp b/dom/media/webrtc/libwebrtcglue/RtpSourceObserver.cpp
new file mode 100644
index 0000000000..f47e89a3c3
--- /dev/null
+++ b/dom/media/webrtc/libwebrtcglue/RtpSourceObserver.cpp
@@ -0,0 +1,197 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=8 sts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "RtpSourceObserver.h"
+#include "nsThreadUtils.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/modules/include/module_common_types.h"
+
+namespace mozilla {
+
+using EntryType = dom::RTCRtpSourceEntryType;
+
+double RtpSourceObserver::RtpSourceEntry::ToLinearAudioLevel() const {
+  // Spec indicates that a value of 127 should be set to 0
+  if (audioLevel == 127) {
+    return 0;
+  }
+  // All other values are calculated as 10^(-rfc_level/20)
+  return std::pow(10, -static_cast<double>(audioLevel) / 20);
+}
+
+RtpSourceObserver::RtpSourceObserver(
+    const dom::RTCStatsTimestampMaker& aTimestampMaker)
+    : mMaxJitterWindow(0), mTimestampMaker(aTimestampMaker) {}
+
+void RtpSourceObserver::OnRtpPacket(const webrtc::RTPHeader& aHeader,
+                                    const uint32_t aJitter) {
+  DOMHighResTimeStamp jsNow = mTimestampMaker.GetNow();
+
+  RefPtr<Runnable> runnable = NS_NewRunnableFunction(
+      "RtpSourceObserver::OnRtpPacket",
+      [this, self = RefPtr<RtpSourceObserver>(this), aHeader, aJitter,
+       jsNow]() {
+        mMaxJitterWindow =
+            std::max(mMaxJitterWindow, static_cast<int64_t>(aJitter) * 2);
+        // We are supposed to report the time at which this packet was played
+        // out, but we have only just received the packet. We try to guess when
+        // it will be played out.
+        // TODO: We need to move where we update these stats to MediaPipeline,
+        // where we send frames to the media track graph. In order to do that,
+        // we will need to have the ssrc (and csrc) for decoded frames, but we
+        // don't have that right now. Once we move this to the correct place, we
+        // will no longer need to keep anything but the most recent data.
+        const auto predictedPlayoutTime = jsNow + aJitter;
+        auto& hist =
+            mRtpSources[GetKey(aHeader.ssrc, EntryType::Synchronization)];
+        hist.Prune(jsNow);
+        // ssrc-audio-level handling
+        hist.Insert(jsNow, predictedPlayoutTime, aHeader.timestamp,
+                    aHeader.extension.hasAudioLevel,
+                    aHeader.extension.audioLevel);
+
+        // csrc-audio-level handling
+        const auto& list = aHeader.extension.csrcAudioLevels;
+        for (uint8_t i = 0; i < aHeader.numCSRCs; i++) {
+          const uint32_t& csrc = aHeader.arrOfCSRCs[i];
+          auto& hist = mRtpSources[GetKey(csrc, EntryType::Contributing)];
+          hist.Prune(jsNow);
+          bool hasLevel = i < list.numAudioLevels;
+          uint8_t level = hasLevel ? list.arrOfAudioLevels[i] : 0;
+          hist.Insert(jsNow, predictedPlayoutTime, aHeader.timestamp, hasLevel,
+                      level);
+        }
+      });
+
+  if (NS_IsMainThread()) {
+    // Code-path for gtests; everything happens on main, and there's no event
+    // loop.
+    runnable->Run();
+  } else {
+    NS_DispatchToMainThread(runnable);
+  }
+}
+
+void RtpSourceObserver::GetRtpSources(
+    nsTArray<dom::RTCRtpSourceEntry>& outSources) const {
+  MOZ_ASSERT(NS_IsMainThread());
+  outSources.Clear();
+  for (const auto& it : mRtpSources) {
+    const RtpSourceEntry* entry =
+        it.second.FindClosestNotAfter(mTimestampMaker.GetNow());
+    if (entry) {
+      dom::RTCRtpSourceEntry domEntry;
+      domEntry.mSource = GetSourceFromKey(it.first);
+      domEntry.mSourceType = GetTypeFromKey(it.first);
+      domEntry.mTimestamp = entry->predictedPlayoutTime;
+      domEntry.mRtpTimestamp = entry->rtpTimestamp;
+      if (entry->hasAudioLevel) {
+        domEntry.mAudioLevel.Construct(entry->ToLinearAudioLevel());
+      }
+      outSources.AppendElement(std::move(domEntry));
+    }
+  }
+}
+
+const RtpSourceObserver::RtpSourceEntry*
+RtpSourceObserver::RtpSourceHistory::FindClosestNotAfter(int64_t aTime) const {
+  MOZ_ASSERT(NS_IsMainThread());
+  // This method scans the history for the entry whose timestamp is closest to a
+  // given timestamp but no greater. Because it is scanning forward, it keeps
+  // track of the closest entry it has found so far in case it overshoots.
+  // There is no before map.begin() which complicates things, so found tracks
+  // if something was really found.
+  auto lastFound = mDetailedHistory.cbegin();
+  bool found = false;
+  for (const auto& it : mDetailedHistory) {
+    if (it.second.predictedPlayoutTime > aTime) {
+      break;
+    }
+    // lastFound can't start before begin, so the first inc must be skipped
+    if (found) {
+      lastFound++;
+    }
+    found = true;
+  }
+  if (found) {
+    return &lastFound->second;
+  }
+  if (HasEvicted() && aTime >= mLatestEviction.predictedPlayoutTime) {
+    return &mLatestEviction;
+  }
+  return nullptr;
+}
+
+void RtpSourceObserver::RtpSourceHistory::Prune(const int64_t aTimeNow) {
+  MOZ_ASSERT(NS_IsMainThread());
+  const auto aTimeT = aTimeNow - mMaxJitterWindow;
+  const auto aTimePrehistory = aTimeNow - kHistoryWindow;
+  bool found = false;
+  // New lower bound of the map
+  auto lower = mDetailedHistory.begin();
+  for (auto& it : mDetailedHistory) {
+    if (it.second.predictedPlayoutTime > aTimeT) {
+      found = true;
+      break;
+    }
+    if (found) {
+      lower++;
+    }
+    found = true;
+  }
+  if (found) {
+    if (lower->second.predictedPlayoutTime > aTimePrehistory) {
+      mLatestEviction = lower->second;
+      mHasEvictedEntry = true;
+    }
+    lower++;
+    mDetailedHistory.erase(mDetailedHistory.begin(), lower);
+  }
+  if (HasEvicted() &&
+      (mLatestEviction.predictedPlayoutTime + kHistoryWindow) < aTimeNow) {
+    mHasEvictedEntry = false;
+  }
+}
+
+void RtpSourceObserver::RtpSourceHistory::Insert(const int64_t aTimeNow,
+                                                 const int64_t aTimestamp,
+                                                 const uint32_t aRtpTimestamp,
+                                                 const bool aHasAudioLevel,
+                                                 const uint8_t aAudioLevel) {
+  MOZ_ASSERT(NS_IsMainThread());
+  Insert(aTimeNow, aTimestamp)
+      .Update(aTimestamp, aRtpTimestamp, aHasAudioLevel, aAudioLevel);
+}
+
+RtpSourceObserver::RtpSourceEntry& RtpSourceObserver::RtpSourceHistory::Insert(
+    const int64_t aTimeNow, const int64_t aTimestamp) {
+  MOZ_ASSERT(NS_IsMainThread());
+  // Time T is the oldest time inside the jitter window (now - jitter)
+  // Time J is the newest time inside the jitter window (now + jitter)
+  // Time x is the jitter adjusted entry time
+  // Time Z is the time of the long term storage element
+  // Times A, B, C are times of entries in the jitter window buffer
+  // x-axis: time
+  // x or x        T   J
+  //  |------Z-----|ABC| -> |------Z-----|ABC|
+  if ((aTimestamp + kHistoryWindow) < aTimeNow ||
+      aTimestamp < mLatestEviction.predictedPlayoutTime) {
+    return mPrehistory;  // A.K.A. /dev/null
+  }
+  mMaxJitterWindow = std::max(mMaxJitterWindow, (aTimestamp - aTimeNow) * 2);
+  const int64_t aTimeT = aTimeNow - mMaxJitterWindow;
+  //           x  T   J
+  // |------Z-----|ABC| -> |--------x---|ABC|
+  if (aTimestamp < aTimeT) {
+    mHasEvictedEntry = true;
+    return mLatestEviction;
+  }
+  //              T  X J
+  // |------Z-----|AB-C| -> |--------x---|ABXC|
+  return mDetailedHistory[aTimestamp];
+}
+
+}  // namespace mozilla