Commit 0fa74e7f authored by Karl Tomlinson's avatar Karl Tomlinson
Browse files

Bug 1804963 get RTPCallerType in AudioContext constructor where window is known non-null r=padenot 

This is a similar approach to that used in PerformanceMainThread.
https://searchfox.org/mozilla-central/rev/2d24d893669ad0fe8d76b0427b25369d35fcc19b/dom/performance/PerformanceMainThread.cpp#99

Differential Revision: https://phabricator.services.mozilla.com/D164518
parent 4819a4a6

dom/media/webaudio/AudioContext.cpp

+3 −5
Original line number Diff line number Diff line
@@ -157,6 +157,7 @@ AudioContext::AudioContext(nsPIDOMWindowInner* aWindow, bool aIsOffline, 
      mSampleRate(GetSampleRateForAudioContext(aIsOffline, aSampleRate)),

      mAudioContextState(AudioContextState::Suspended),

      mNumberOfChannels(aNumberOfChannels),

      mRTPCallerType(aWindow->AsGlobal()->GetRTPCallerType()),

      mIsOffline(aIsOffline),

      mIsStarted(!aIsOffline),

      mIsShutDown(false),
@@ -739,24 +740,21 @@ double AudioContext::CurrentTime() { 


  double rawTime = track->TrackTimeToSeconds(track->GetCurrentTime());



  RTPCallerType callerType = GetParentObject()->AsGlobal()->GetRTPCallerType();



  // 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(callerType) / 1000.0) {

      nsRFPService::TimerResolution(mRTPCallerType) / 1000.0) {

    return rawTime;

  }



  MOZ_ASSERT(GetParentObject()->AsGlobal());

  // The value of a MediaTrack'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(

      rawTime, GetRandomTimelineSeed(), callerType);

      rawTime, GetRandomTimelineSeed(), mRTPCallerType);

}



nsISerialEventTarget* AudioContext::GetMainThread() const {

dom/media/webaudio/AudioContext.h

+1 −0
Original line number Diff line number Diff line
@@ -420,6 +420,7 @@ class AudioContext final : public DOMEventTargetHelper, 
  RefPtr<BasicWaveFormCache> mBasicWaveFormCache;

  // Number of channels passed in the OfflineAudioContext ctor.

  uint32_t mNumberOfChannels;

  const RTPCallerType mRTPCallerType;

  const bool mIsOffline;

  // true iff realtime or startRendering() has been called.

  bool mIsStarted;

testing/web-platform/tests/webaudio/the-audio-api/the-audiocontext-interface/crashtests/currentTime-after-discard.html

0 → 100644
+14 −0
Original line number Diff line number Diff line 
<html>

<head>

  <title>

    Test currentTime after browsing context discard

  </title>

</head>

<script>

  const frame = document.createElement('frame');

  document.documentElement.appendChild(frame);

  const ctx = new frame.contentWindow.AudioContext();

  frame.remove();

  ctx.currentTime;

</script>

</html>