1 files changed, 151 insertions, 0 deletions
diff --git a/dom/media/webrtc/libwebrtcglue/MediaConduitInterface.cpp b/dom/media/webrtc/libwebrtcglue/MediaConduitInterface.cpp
new file mode 100644
index 0000000000..c0718739e9
--- /dev/null
+++ b/dom/media/webrtc/libwebrtcglue/MediaConduitInterface.cpp
@@ -0,0 +1,151 @@
+/* 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 "MediaConduitInterface.h"
+
+#include "nsTArray.h"
+#include "mozilla/Assertions.h"
+#include "MainThreadUtils.h"
+#include "SystemTime.h"
+
+#include "system_wrappers/include/clock.h"
+
+namespace mozilla {
+
+void MediaSessionConduit::GetRtpSources(
+    nsTArray<dom::RTCRtpSourceEntry>& outSources) const {
+  MOZ_ASSERT(NS_IsMainThread());
+  if (mSourcesUpdateNeeded) {
+    UpdateRtpSources(GetUpstreamRtpSources());
+    OnSourcesUpdated();
+  }
+  outSources.Clear();
+  for (auto& [key, entry] : mSourcesCache) {
+    (void)key;
+    outSources.AppendElement(entry);
+  }
+
+  struct TimestampComparator {
+    bool LessThan(const dom::RTCRtpSourceEntry& aLhs,
+                  const dom::RTCRtpSourceEntry& aRhs) const {
+      // Sort descending!
+      return aLhs.mTimestamp > aRhs.mTimestamp;
+    }
+
+    bool Equals(const dom::RTCRtpSourceEntry& aLhs,
+                const dom::RTCRtpSourceEntry& aRhs) const {
+      return aLhs.mTimestamp == aRhs.mTimestamp;
+    }
+  };
+
+  // *sigh* We have to re-sort this by JS timestamp; we can run into cases
+  // where the libwebrtc timestamps are not in exactly the same order as JS
+  // timestamps due to clock differences (wibbly-wobbly, timey-wimey stuff)
+  outSources.Sort(TimestampComparator());
+}
+
+static double rtpToDomAudioLevel(uint8_t aAudioLevel) {
+  if (aAudioLevel == 127) {
+    // Spec indicates that a value of 127 should be set to 0
+    return 0;
+  }
+
+  // All other values are calculated as 10^(-rfc_level/20)
+  return std::pow(10, -aAudioLevel / 20.0);
+}
+
+void MediaSessionConduit::UpdateRtpSources(
+    const std::vector<webrtc::RtpSource>& aSources) const {
+  MOZ_ASSERT(NS_IsMainThread());
+  // Empty out the cache; we'll copy things back as needed
+  auto cache = std::move(mSourcesCache);
+
+  for (const auto& source : aSources) {
+    SourceKey key(source);
+    auto it = cache.find(key);
+    if (it != cache.end()) {
+      // This source entry was already in the cache, and should continue to be
+      // present in exactly the same form as before. This means we do _not_
+      // want to perform the timestamp adjustment again, since it might yield a
+      // slightly different result. This is why we copy this entry from the old
+      // cache instead of simply rebuilding it, and is also why we key the
+      // cache based on timestamp (keying the cache based on timestamp also
+      // gets us the ordering we want, conveniently).
+      mSourcesCache[key] = it->second;
+      continue;
+    }
+
+    // This is something we did not already have in the cache.
+    dom::RTCRtpSourceEntry domEntry;
+    domEntry.mSource = source.source_id();
+    switch (source.source_type()) {
+      case webrtc::RtpSourceType::SSRC:
+        domEntry.mSourceType = dom::RTCRtpSourceEntryType::Synchronization;
+        break;
+      case webrtc::RtpSourceType::CSRC:
+        domEntry.mSourceType = dom::RTCRtpSourceEntryType::Contributing;
+        break;
+      default:
+        MOZ_CRASH("Unexpected RTCRtpSourceEntryType");
+    }
+
+    if (source.audio_level()) {
+      domEntry.mAudioLevel.Construct(rtpToDomAudioLevel(*source.audio_level()));
+    }
+
+    // These timestamps are always **rounded** to milliseconds. That means they
+    // can jump up to half a millisecond into the future. We compensate for that
+    // here so that things seem consistent to js.
+    domEntry.mTimestamp = GetTimestampMaker().ReduceRealtimePrecision(
+        webrtc::Timestamp::Millis(source.timestamp_ms()) -
+        webrtc::TimeDelta::Micros(500));
+    domEntry.mRtpTimestamp = source.rtp_timestamp();
+    mSourcesCache[key] = domEntry;
+  }
+}
+
+void MediaSessionConduit::OnSourcesUpdated() const {
+  MOZ_ASSERT(NS_IsMainThread());
+  MOZ_ASSERT(mSourcesUpdateNeeded);
+  mSourcesUpdateNeeded = false;
+  // Reset the updateNeeded flag and clear the cache in a direct task, i.e.,
+  // as soon as the current task has finished.
+  AbstractThread::GetCurrent()->TailDispatcher().AddDirectTask(
+      NS_NewRunnableFunction(
+          __func__, [this, self = RefPtr<const MediaSessionConduit>(this)] {
+            mSourcesUpdateNeeded = true;
+            mSourcesCache.clear();
+          }));
+}
+
+void MediaSessionConduit::InsertAudioLevelForContributingSource(
+    const uint32_t aCsrcSource, const int64_t aTimestamp,
+    const uint32_t aRtpTimestamp, const bool aHasAudioLevel,
+    const uint8_t aAudioLevel) {
+  MOZ_ASSERT(NS_IsMainThread());
+
+  if (mSourcesUpdateNeeded) {
+    OnSourcesUpdated();
+  }
+
+  dom::RTCRtpSourceEntry domEntry;
+  domEntry.mSource = aCsrcSource;
+  domEntry.mSourceType = dom::RTCRtpSourceEntryType::Contributing;
+  domEntry.mTimestamp = aTimestamp;
+  domEntry.mRtpTimestamp = aRtpTimestamp;
+  if (aHasAudioLevel) {
+    domEntry.mAudioLevel.Construct(rtpToDomAudioLevel(aAudioLevel));
+  }
+
+  webrtc::Timestamp libwebrtcNow = GetTimestampMaker().GetNowRealtime();
+  double jsNow = GetTimestampMaker().ReduceRealtimePrecision(libwebrtcNow);
+  double ago = jsNow - aTimestamp;
+  webrtc::Timestamp convertedTimestamp =
+      libwebrtcNow - webrtc::TimeDelta::Millis(ago);
+
+  SourceKey key(convertedTimestamp.ms<uint32_t>(), aCsrcSource);
+  mSourcesCache[key] = domEntry;
+}
+
+}  // namespace mozilla