Bug 1446346 Do not clamp or jitter the AudioContext's CurrentTime if its interval is larger than our precision. r=jib

MozReview-Commit-ID: Bc1cto3pBKL

--HG--
extra : rebase_source : ace58df9f1669beacc176ab968f45a09ca44be2e

browser/components/resistfingerprinting/test/mochitest/test_reduce_time_precision.html

@@ -175,6 +175,16 @@ https://trac.torproject.org/projects/tor/ticket/1517
     // and check if it is rounded
     for (let timeStampCode of timeStampCodesDOM) {
       let timeStamp = eval(timeStampCode);
+
+      // Audio Contexts increment in intervals of (minimum) 5.4ms, so we don't
+      // clamp/jitter if the timer precision is les than that.
+      // (Technically on MBPs they increment in intervals of 2.6 but this is
+      // non-standard and will eventually be changed. We don't cover this situation
+      // because we don't really support arbitrary Timer Precision, especially in
+      // the 2.6 - 5.4ms interval.)
+      if (timeStampCode.includes("audioContext") && expectedPrecision < 5.4)
+        continue;
+
       ok(isRounded(timeStamp, expectedPrecision),
         "pref: " + prefname + " - '" +
          "'" + timeStampCode +

dom/media/webaudio/AudioContext.cpp

@@ -668,17 +668,22 @@ AudioContext::CurrentTime()
 {
   MediaStream* stream = Destination()->Stream();
 
-  if (!mIsStarted &&
-    stream->StreamTimeToSeconds(stream->GetCurrentTime()) == 0) {
-      return 0;
+  double rawTime = stream->StreamTimeToSeconds(stream->GetCurrentTime());
+
+  // CurrentTime increments in intervals of 128/sampleRate. If the Timer
+  // Precision Reduction is smaller than this interval, the jittered time
+  // can always be reversed to the raw step of the interval. In that case
+  // we can simply return the un-reduced time; and avoid breaking tests.
+  // We have to convert each variable into a common magnitude, we choose ms.
+  if ((128/mSampleRate) * 1000.0 > nsRFPService::TimerResolution() / 1000.0) {
+    return rawTime;
   }
 
   // The value of a MediaStream's CurrentTime will always advance forward; it will never
   // reset (even if one rewinds a video.) Therefore we can use a single Random Seed
   // initialized at the same time as the object.
   return nsRFPService::ReduceTimePrecisionAsSecs(
-    stream->StreamTimeToSeconds(stream->GetCurrentTime()),
-    GetRandomTimelineSeed());
+    rawTime, GetRandomTimelineSeed());
 }
 
 void AudioContext::DisconnectFromOwner()

toolkit/components/resistfingerprinting/nsRFPService.cpp

@@ -107,8 +107,9 @@ nsRFPService::GetOrCreate()
   return sRFPService;
 }
 
-inline double
-TimerResolution()
+/* static */
+double
+nsRFPService::TimerResolution()
 {
   if(nsRFPService::IsResistFingerprintingEnabled()) {
     return max(100000.0, (double)sResolutionUSec);

toolkit/components/resistfingerprinting/nsRFPService.h

@@ -162,6 +162,7 @@ public:
   static nsRFPService* GetOrCreate();
   static bool IsResistFingerprintingEnabled();
   static bool IsTimerPrecisionReductionEnabled(TimerPrecisionType aType);
+  static double TimerResolution();
 
   enum TimeScale {
     Seconds      = 1,