Bug 1419093 - P2 - update rtp source impl & unit tests r=mjf

Update the c++ side with the new WebIDL definitions, note that
many fields are now required. See P1 for more details.


MozReview-Commit-ID: FKHi86Nj7UO

--HG--
extra : rebase_source : 7491551b46a9fb0e8c8cba77861b01f90ea95644

media/webrtc/signaling/gtest/rtpsources_unittests.cpp

@@ -265,14 +265,14 @@ public:
     bool ssrcFound = false;
     bool csrcFound = true;
     for (auto& entry : outLevels) {
-      if (entry.mSource.Value() == ssrc) {
+      if (entry.mSource == ssrc) {
         ssrcFound = true;
-        EXPECT_EQ(entry.mSourceType.Value(),
+        EXPECT_EQ(entry.mSourceType,
                   dom::RTCRtpSourceEntryType::Synchronization);
       }
-      if (entry.mSource.Value() == csrc) {
+      if (entry.mSource == csrc) {
         csrcFound = true;
-        EXPECT_EQ(entry.mSourceType.Value(),
+        EXPECT_EQ(entry.mSourceType,
                   dom::RTCRtpSourceEntryType::Contributing);
       }
     }
@@ -305,19 +305,19 @@ public:
     bool csrc0Found = true;
     bool csrc1Found = true;
     for (auto& entry : outLevels) {
-      if (entry.mSource.Value() == ssrc) {
+      if (entry.mSource == ssrc) {
         ssrcFound = true;
-        EXPECT_EQ(entry.mSourceType.Value(),
+        EXPECT_EQ(entry.mSourceType,
                   dom::RTCRtpSourceEntryType::Synchronization);
       }
-      if (entry.mSource.Value() == csrc0) {
+      if (entry.mSource == csrc0) {
         csrc0Found = true;
-        EXPECT_EQ(entry.mSourceType.Value(),
+        EXPECT_EQ(entry.mSourceType,
                   dom::RTCRtpSourceEntryType::Contributing);
       }
-      if (entry.mSource.Value() == csrc1) {
+      if (entry.mSource == csrc1) {
         csrc1Found = true;
-        EXPECT_EQ(entry.mSourceType.Value(),
+        EXPECT_EQ(entry.mSourceType,
                   dom::RTCRtpSourceEntryType::Contributing);
       }
     }

media/webrtc/signaling/src/media-conduit/RtpSourceObserver.cpp

@@ -7,6 +7,17 @@ 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() :
   mLevelGuard("RtpSourceObserver::mLevelGuard") {}
 
@@ -51,11 +62,11 @@ RtpSourceObserver::GetRtpSources(const int64_t aTimeNow,
     const RtpSourceEntry* entry = it.second.FindClosestNotAfter(aTimeNow);
     if (entry) {
       dom::RTCRtpSourceEntry domEntry;
-      domEntry.mSource.Construct(GetSourceFromKey(it.first));
-      domEntry.mSourceType.Construct(GetTypeFromKey(it.first));
-      domEntry.mTimestamp.Construct(entry->jitterAdjustedTimestamp);
+      domEntry.mSource = GetSourceFromKey(it.first);
+      domEntry.mSourceType = GetTypeFromKey(it.first);
+      domEntry.mTimestamp = entry->jitterAdjustedTimestamp;
       if (entry->hasAudioLevel) {
-        domEntry.mAudioLevel.Construct(entry->audioLevel);
+        domEntry.mAudioLevel.Construct(entry->ToLinearAudioLevel());
       }
       outSources.AppendElement(std::move(domEntry));
     }

media/webrtc/signaling/src/media-conduit/RtpSourceObserver.h

@@ -62,6 +62,9 @@ private:
       hasAudioLevel = aHasAudioLevel && !(aAudioLevel & 0x80);
       audioLevel = aAudioLevel;
     }
+    // Sets the audio level nullable according to the linear scale
+    // outlined in the webrtc-pc spec.
+    double ToLinearAudioLevel() const;
     // Time this information was received + jitter
     int64_t jitterAdjustedTimestamp = 0;
     bool hasAudioLevel = false;