Backed out 2 changesets (bug 1754006) for mda failures on test_bug1113600.html CLOSED TREE

Backed out changeset f22af4b15704 (bug 1754006)
Backed out changeset 078521a5c82b (bug 1754006)

dom/media/mediasink/AudioSink.cpp

@@ -16,7 +16,6 @@
 #include "mozilla/StaticPrefs_media.h"
 #include "mozilla/StaticPrefs_dom.h"
 #include "nsPrintfCString.h"
-#include "Tracing.h"
 
 namespace mozilla {
 
@@ -31,6 +30,9 @@ extern LazyLogModule gMediaDecoderLog;
 // The amount of audio frames that is used to fuzz rounding errors.
 static const int64_t AUDIO_FUZZ_FRAMES = 1;
 
+// Amount of audio frames we will be processing ahead of use
+static const int32_t LOW_AUDIO_USECS = 300000;
+
 using media::TimeUnit;
 
 AudioSink::AudioSink(AbstractThread* aThread,
@@ -52,11 +54,9 @@ AudioSink::AudioSink(AbstractThread* aThread,
           StaticPrefs::dom_media_silence_duration_for_audibility()),
       mIsAudioDataAudible(false),
       mProcessedQueueFinished(false),
-      mAudioQueue(aAudioQueue),
-      mProcessedQueueThresholdMS(
-          StaticPrefs::media_audio_audiosink_threshold_ms()) {
+      mAudioQueue(aAudioQueue) {
   // Twice the limit that trigger a refill.
-  float capacitySeconds = mProcessedQueueThresholdMS / 1000.f * 2;
+  float capacitySeconds = LOW_AUDIO_USECS / 1000. / 1000. * 2;
   mProcessedSPSCQueue =
       MakeUnique<SPSCQueue<AudioDataValue>>(static_cast<uint32_t>(
           capacitySeconds * static_cast<float>(mOutputChannels * mOutputRate)));
@@ -216,11 +216,6 @@ uint32_t AudioSink::PopFrames(AudioDataValue* aBuffer, uint32_t aFrames,
   if (aAudioThreadChanged) {
     mProcessedSPSCQueue->ResetThreadIds();
   }
-
-  TRACE_COMMENT("AudioSink::PopFrames", "%u frames (ringbuffer: %u/%u)",
-                aFrames, SampleToFrame(mProcessedSPSCQueue->AvailableRead()),
-                SampleToFrame(mProcessedSPSCQueue->Capacity()));
-
   const int samplesToPop = static_cast<int>(aFrames * mOutputChannels);
   const int samplesRead = mProcessedSPSCQueue->Dequeue(aBuffer, samplesToPop);
   MOZ_ASSERT(samplesRead % mOutputChannels == 0);
@@ -230,8 +225,6 @@ uint32_t AudioSink::PopFrames(AudioDataValue* aBuffer, uint32_t aFrames,
       SINK_LOG("Last PopFrames -- Source ended.");
     } else {
       NS_WARNING("Underrun when popping samples from audiosink ring buffer.");
-      TRACE_COMMENT("AudioSink::PopFrames", "Underrun %u frames missing",
-                    SampleToFrame(samplesToPop - samplesRead));
     }
     // silence the rest
     PodZero(aBuffer + samplesRead, samplesToPop - samplesRead);
@@ -275,9 +268,10 @@ void AudioSink::OnAudioPushed(const RefPtr<AudioData>& aSample) {
   NotifyAudioNeeded();
 }
 
-uint32_t AudioSink::AudioQueuedInRingBufferMS() const {
+uint32_t AudioSink::AudioQueuedInRingBufferUs() const {
   return static_cast<uint32_t>(
-      1000 * SampleToFrame(mProcessedSPSCQueue->AvailableRead()) / mOutputRate);
+      1000. * 1000 * SampleToFrame(mProcessedSPSCQueue->AvailableRead()) /
+      mOutputRate);
 }
 
 uint32_t AudioSink::SampleToFrame(uint32_t aSamples) const {
@@ -289,7 +283,7 @@ void AudioSink::NotifyAudioNeeded() {
              "Not called from the owner's thread");
 
   while (mAudioQueue.GetSize() &&
-         AudioQueuedInRingBufferMS() < mProcessedQueueThresholdMS) {
+         AudioQueuedInRingBufferUs() < LOW_AUDIO_USECS) {
     // Check if there's room in our ring buffer.
     if (mAudioQueue.PeekFront()->Frames() >
         SampleToFrame(mProcessedSPSCQueue->AvailableWrite())) {
@@ -398,7 +392,6 @@ void AudioSink::NotifyAudioNeeded() {
       } else {
         silenceData = CreateAudioFromBuffer(std::move(silenceBuffer), data);
       }
-      TRACE("Pushing silence");
       PushProcessedAudio(silenceData);
     }
 
@@ -428,10 +421,6 @@ uint32_t AudioSink::PushProcessedAudio(AudioData* aData) {
     return 0;
   }
   int framesToEnqueue = static_cast<int>(aData->Frames() * aData->mChannels);
-  TRACE_COMMENT("AudioSink::PushProcessedAudio", "%u frames (%u/%u)",
-                framesToEnqueue,
-                SampleToFrame(mProcessedSPSCQueue->AvailableWrite()),
-                SampleToFrame(mProcessedSPSCQueue->Capacity()));
   DebugOnly<int> rv =
       mProcessedSPSCQueue->Enqueue(aData->Data().Elements(), framesToEnqueue);
   NS_WARNING_ASSERTION(

dom/media/mediasink/AudioSink.h

@@ -129,7 +129,7 @@ class AudioSink : private AudioStream::DataSource {
       AlignedAudioBuffer&& aBuffer, AudioData* aReference);
   // Add data to the processsed queue return the number of frames added.
   uint32_t PushProcessedAudio(AudioData* aData);
-  uint32_t AudioQueuedInRingBufferMS() const;
+  uint32_t AudioQueuedInRingBufferUs() const;
   uint32_t SampleToFrame(uint32_t aSamples) const;
   UniquePtr<AudioConverter> mConverter;
   UniquePtr<SPSCQueue<AudioDataValue>> mProcessedSPSCQueue;
@@ -153,7 +153,6 @@ class AudioSink : private AudioStream::DataSource {
 
   Atomic<bool> mProcessedQueueFinished;
   MediaQueue<AudioData>& mAudioQueue;
-  const float mProcessedQueueThresholdMS;
 };
 
 }  // namespace mozilla

modules/libpref/init/StaticPrefList.yaml

@@ -9512,14 +9512,6 @@
   value: 8
   mirror: always
 
-# The duration, in milliseconds, of decoded audio to keep around in the
-# AudioSink ring-buffer. New decoding operations are started when this limit is
-# reached. The total size of the ring-buffer is slightly bigger than this.
-- name: media.audio.audiosink.threshold_ms
-  type: AtomicFloat
-  value: 1000.0
-  mirror: always
-
 
 #---------------------------------------------------------------------------
 # Prefs starting with "midi."