Bug 1307042: generate fake audio for getUserMedia from MSG callbacks r=pehrsons

dom/media/webrtc/MediaEngineDefault.cpp

@@ -13,6 +13,7 @@
 #include "ImageTypes.h"
 #include "prmem.h"
 #include "nsContentUtils.h"
+#include "MediaStreamGraph.h"
 
 #include "nsIFilePicker.h"
 #include "nsIPrefService.h"
@@ -367,14 +368,13 @@ private:
 /**
  * Default audio source.
  */
-NS_IMPL_ISUPPORTS(MediaEngineDefaultAudioSource, nsITimerCallback)
+
+NS_IMPL_ISUPPORTS0(MediaEngineDefaultAudioSource)
 
 MediaEngineDefaultAudioSource::MediaEngineDefaultAudioSource()
   : MediaEngineAudioSource(kReleased)
-  , mPrincipalHandle(PRINCIPAL_HANDLE_NONE)
+  , mLastNotify(0)
-  , mTimer(nullptr)
+{}
-{
-}
 
 MediaEngineDefaultAudioSource::~MediaEngineDefaultAudioSource()
 {}
@@ -453,43 +453,15 @@ MediaEngineDefaultAudioSource::Start(SourceMediaStream* aStream, TrackID aID,
     return NS_ERROR_FAILURE;
   }
 
-  mTimer = do_CreateInstance(NS_TIMER_CONTRACTID);
-  if (!mTimer) {
-    return NS_ERROR_FAILURE;
-  }
-
-  mSource = aStream;
-
-  // We try to keep the appended data at this size.
-  // Make it two timer intervals to try to avoid underruns.
-  mBufferSize = 2 * (AUDIO_RATE * DEFAULT_AUDIO_TIMER_MS) / 1000;
-
   // AddTrack will take ownership of segment
   AudioSegment* segment = new AudioSegment();
-  AppendToSegment(*segment, mBufferSize);
+  aStream->AddAudioTrack(aID, AUDIO_RATE, 0, segment, SourceMediaStream::ADDTRACK_QUEUED);
-  mSource->AddAudioTrack(aID, AUDIO_RATE, 0, segment, SourceMediaStream::ADDTRACK_QUEUED);
 
   // Remember TrackID so we can finish later
   mTrackID = aID;
 
-  // Remember PrincipalHandle since we don't append in NotifyPull.
+  mLastNotify = 0;
-  mPrincipalHandle = aPrincipalHandle;
-
-  mLastNotify = TimeStamp::Now();
-
-  // 1 Audio frame per 10ms
-  // We'd like to do this for Android Debug as well, but that breaks tests that check for
-  // audio frequency data.
-#if defined(MOZ_WIDGET_GONK) && defined(DEBUG)
-// emulator debug is very, very slow and has problems dealing with realtime audio inputs
-  mTimer->InitWithCallback(this, DEFAULT_AUDIO_TIMER_MS*10,
-                           nsITimer::TYPE_REPEATING_PRECISE_CAN_SKIP);
-#else
-  mTimer->InitWithCallback(this, DEFAULT_AUDIO_TIMER_MS,
-                           nsITimer::TYPE_REPEATING_PRECISE_CAN_SKIP);
-#endif
   mState = kStarted;
-
   return NS_OK;
 }
 
@@ -499,13 +471,6 @@ MediaEngineDefaultAudioSource::Stop(SourceMediaStream *aSource, TrackID aID)
   if (mState != kStarted) {
     return NS_ERROR_FAILURE;
   }
-  if (!mTimer) {
-    return NS_ERROR_FAILURE;
-  }
-
-  mTimer->Cancel();
-  mTimer = nullptr;
-
   aSource->EndTrack(aID);
 
   mState = kStopped;
@@ -524,7 +489,8 @@ MediaEngineDefaultAudioSource::Restart(AllocationHandle* aHandle,
 
 void
 MediaEngineDefaultAudioSource::AppendToSegment(AudioSegment& aSegment,
-                                               TrackTicks aSamples)
+                                               TrackTicks aSamples,
+                                               const PrincipalHandle& aPrincipalHandle)
 {
   RefPtr<SharedBuffer> buffer = SharedBuffer::Create(aSamples * sizeof(int16_t));
   int16_t* dest = static_cast<int16_t*>(buffer->Data());
@@ -532,28 +498,24 @@ MediaEngineDefaultAudioSource::AppendToSegment(AudioSegment& aSegment,
   mSineGenerator->generate(dest, aSamples);
   AutoTArray<const int16_t*,1> channels;
   channels.AppendElement(dest);
-  aSegment.AppendFrames(buffer.forget(), channels, aSamples, mPrincipalHandle);
+  aSegment.AppendFrames(buffer.forget(), channels, aSamples, aPrincipalHandle);
 }
 
-NS_IMETHODIMP
+void
-MediaEngineDefaultAudioSource::Notify(nsITimer* aTimer)
+MediaEngineDefaultAudioSource::NotifyPull(MediaStreamGraph* aGraph,
+                                          SourceMediaStream *aSource,
+                                          TrackID aID,
+                                          StreamTime aDesiredTime,
+                                          const PrincipalHandle& aPrincipalHandle)
 {
-  TimeStamp now = TimeStamp::Now();
+  MOZ_ASSERT(aID == mTrackID);
-  TimeDuration timeSinceLastNotify = now - mLastNotify;
-  mLastNotify = now;
-  TrackTicks samplesSinceLastNotify =
-    RateConvertTicksRoundUp(AUDIO_RATE, 1000000, timeSinceLastNotify.ToMicroseconds());
-
-  // If it's been longer since the last Notify() than mBufferSize holds, we
-  // have underrun and the MSG had to append silence while waiting for us
-  // to push more data. In this case we reset to mBufferSize again.
-  TrackTicks samplesToAppend = std::min(samplesSinceLastNotify, mBufferSize);
-
   AudioSegment segment;
-  AppendToSegment(segment, samplesToAppend);
+  // avoid accumulating rounding errors
-  mSource->AppendToTrack(mTrackID, &segment);
+  TrackTicks desired = aSource->TimeToTicksRoundUp(AUDIO_RATE, aDesiredTime);
-
+  TrackTicks delta = desired - mLastNotify;
-  return NS_OK;
+  mLastNotify += delta;
+  AppendToSegment(segment, delta, aPrincipalHandle);
+  aSource->AppendToTrack(mTrackID, &segment);
 }
 
 void

dom/media/webrtc/MediaEngineDefault.h

@@ -97,8 +97,7 @@ protected:
   nsCOMPtr<nsITimer> mTimer;
   // mMonitor protects mImage access/changes, and transitions of mState
   // from kStarted to kStopped (which are combined with EndTrack() and
-  // image changes).  Note that mSources is not accessed from other threads
+  // image changes).
-  // for video and is not protected.
   Monitor mMonitor;
   RefPtr<layers::Image> mImage;
 
@@ -111,8 +110,7 @@ protected:
 
 class SineWaveGenerator;
 
-class MediaEngineDefaultAudioSource : public nsITimerCallback,
+class MediaEngineDefaultAudioSource : public MediaEngineAudioSource
-                                      public MediaEngineAudioSource
 {
 public:
   MediaEngineDefaultAudioSource();
@@ -135,22 +133,14 @@ public:
                    const nsString& aDeviceId,
                    const char** aOutBadConstraint) override;
   void SetDirectListeners(bool aHasDirectListeners) override {};
-  void AppendToSegment(AudioSegment& aSegment,
+  void inline AppendToSegment(AudioSegment& aSegment,
-                       TrackTicks aSamples);
+                              TrackTicks aSamples,
+                              const PrincipalHandle& aPrincipalHandle);
   void NotifyPull(MediaStreamGraph* aGraph,
                   SourceMediaStream *aSource,
                   TrackID aId,
                   StreamTime aDesiredTime,
-                  const PrincipalHandle& aPrincipalHandle) override
+                  const PrincipalHandle& aPrincipalHandle) override;
-  {
-#ifdef DEBUG
-    StreamTracks::Track* data = aSource->FindTrack(aId);
-    NS_WARNING_ASSERTION(
-      !data || data->IsEnded() ||
-      aDesiredTime <= aSource->GetEndOfAppendedData(aId),
-      "MediaEngineDefaultAudioSource data underrun");
-#endif
-  }
 
   void NotifyOutputData(MediaStreamGraph* aGraph,
                         AudioDataValue* aBuffer, size_t aFrames,
@@ -180,19 +170,15 @@ public:
       const nsString& aDeviceId) const override;
 
   NS_DECL_THREADSAFE_ISUPPORTS
-  NS_DECL_NSITIMERCALLBACK
 
 protected:
   ~MediaEngineDefaultAudioSource();
 
   TrackID mTrackID;
-  PrincipalHandle mPrincipalHandle;
-  nsCOMPtr<nsITimer> mTimer;
 
-  TimeStamp mLastNotify;
+  TrackTicks mLastNotify; // Accessed in ::Start(), then on NotifyPull (from MSG thread)
-  TrackTicks mBufferSize;
 
-  SourceMediaStream* mSource;
+  // Created on Allocate, then accessed from NotifyPull (MSG thread)
   nsAutoPtr<SineWaveGenerator> mSineGenerator;
 };
 

media/webrtc/signaling/test/FakeMediaStreams.h

@@ -238,6 +238,8 @@ class Fake_MediaStream {
   double StreamTimeToSeconds(mozilla::StreamTime aTime);
   mozilla::StreamTime
   TicksToTimeRoundDown(mozilla::TrackRate aRate, mozilla::TrackTicks aTicks);
+  mozilla::TrackTicks TimeToTicksRoundUp(mozilla::TrackRate aRate,
+                                         mozilla::StreamTime aTime);
 
   NS_INLINE_DECL_THREADSAFE_REFCOUNTING(Fake_MediaStream);
 

media/webrtc/signaling/test/FakeMediaStreamsImpl.h

@@ -24,6 +24,11 @@ double Fake_MediaStream::StreamTimeToSeconds(mozilla::StreamTime aTime) {
   return static_cast<double>(aTime)/GRAPH_RATE;
 }
 
+mozilla::TrackTicks Fake_MediaStream::TimeToTicksRoundUp(mozilla::TrackRate aRate,
+                                                         mozilla::StreamTime aTime) {
+  return (aTime * aRate) / GRAPH_RATE;
+}
+
 mozilla::StreamTime
 Fake_MediaStream::TicksToTimeRoundDown(mozilla::TrackRate aRate,
                                        mozilla::TrackTicks aTicks) {