Bug 1754006 - Increase the processed media queue size and threshold in AudioSink. r=alwu

Differential Revision: https://phabricator.services.mozilla.com/D138028

dom/media/mediasink/AudioSink.cpp

@@ -30,9 +30,6 @@ 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,
@@ -54,9 +51,11 @@ AudioSink::AudioSink(AbstractThread* aThread,
           StaticPrefs::dom_media_silence_duration_for_audibility()),
       mIsAudioDataAudible(false),
       mProcessedQueueFinished(false),
-      mAudioQueue(aAudioQueue) {
+      mAudioQueue(aAudioQueue),
+      mProcessedQueueThresholdMS(
+          StaticPrefs::media_audio_audiosink_threshold_ms()) {
   // Twice the limit that trigger a refill.
-  float capacitySeconds = LOW_AUDIO_USECS / 1000. / 1000. * 2;
+  float capacitySeconds = mProcessedQueueThresholdMS / 1000.f * 2;
   mProcessedSPSCQueue =
       MakeUnique<SPSCQueue<AudioDataValue>>(static_cast<uint32_t>(
           capacitySeconds * static_cast<float>(mOutputChannels * mOutputRate)));
@@ -268,10 +267,9 @@ void AudioSink::OnAudioPushed(const RefPtr<AudioData>& aSample) {
   NotifyAudioNeeded();
 }
 
-uint32_t AudioSink::AudioQueuedInRingBufferUs() const {
+uint32_t AudioSink::AudioQueuedInRingBufferMS() const {
   return static_cast<uint32_t>(
-      1000. * 1000 * SampleToFrame(mProcessedSPSCQueue->AvailableRead()) /
-      mOutputRate);
+      1000 * SampleToFrame(mProcessedSPSCQueue->AvailableRead()) / mOutputRate);
 }
 
 uint32_t AudioSink::SampleToFrame(uint32_t aSamples) const {
@@ -283,7 +281,7 @@ void AudioSink::NotifyAudioNeeded() {
              "Not called from the owner's thread");
 
   while (mAudioQueue.GetSize() &&
-         AudioQueuedInRingBufferUs() < LOW_AUDIO_USECS) {
+         AudioQueuedInRingBufferMS() < mProcessedQueueThresholdMS) {
     // Check if there's room in our ring buffer.
     if (mAudioQueue.PeekFront()->Frames() >
         SampleToFrame(mProcessedSPSCQueue->AvailableWrite())) {

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 AudioQueuedInRingBufferUs() const;
+  uint32_t AudioQueuedInRingBufferMS() const;
   uint32_t SampleToFrame(uint32_t aSamples) const;
   UniquePtr<AudioConverter> mConverter;
   UniquePtr<SPSCQueue<AudioDataValue>> mProcessedSPSCQueue;
@@ -153,6 +153,7 @@ class AudioSink : private AudioStream::DataSource {
 
   Atomic<bool> mProcessedQueueFinished;
   MediaQueue<AudioData>& mAudioQueue;
+  const float mProcessedQueueThresholdMS;
 };
 
 }  // namespace mozilla

modules/libpref/init/StaticPrefList.yaml

@@ -9517,6 +9517,14 @@
   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."