Commit 0541c7f3 authored by Narcis Beleuzu's avatar Narcis Beleuzu
Browse files

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)
parent e81e5f0c

dom/media/mediasink/AudioSink.cpp

+9 −20
Original line number Diff line number Diff line
@@ -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

+1 −2
Original line number Diff line number Diff line
@@ -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

+0 −8
Original line number Diff line number Diff line
@@ -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."