diff --git a/dom/media/MediaDecoderStateMachine.cpp b/dom/media/MediaDecoderStateMachine.cpp index 5b688778bf39..1270056b36ed 100644 --- a/dom/media/MediaDecoderStateMachine.cpp +++ b/dom/media/MediaDecoderStateMachine.cpp @@ -88,15 +88,15 @@ using namespace mozilla::media; // constants directly, so we put them in a namespace. namespace detail { -// If audio queue has less than this many usecs of decoded audio, we won't risk +// If audio queue has less than this much decoded audio, we won't risk // trying to decode the video, we'll skip decoding video up to the next // keyframe. We may increase this value for an individual decoder if we // encounter video frames which take a long time to decode. -static constexpr auto LOW_AUDIO_USECS = TimeUnit::FromMicroseconds(300000); +static constexpr auto LOW_AUDIO_THRESHOLD = TimeUnit::FromMicroseconds(300000); // If more than this many usecs of decoded audio is queued, we'll hold off // decoding more audio. If we increase the low audio threshold (see -// LOW_AUDIO_USECS above) we'll also increase this value to ensure it's not +// LOW_AUDIO_THRESHOLD above) we'll also increase this value to ensure it's not // less than the low audio threshold. static const int64_t AMPLE_AUDIO_USECS = 2000000; @@ -116,7 +116,7 @@ static const int32_t LOW_VIDEO_THRESHOLD_USECS = 60000; // Arbitrary "frame duration" when playing only audio. static const int AUDIO_DURATION_USECS = 40000; -// If we increase our "low audio threshold" (see LOW_AUDIO_USECS above), we +// If we increase our "low audio threshold" (see LOW_AUDIO_THRESHOLD above), we // use this as a factor in all our calculations. Increasing this will cause // us to be more likely to increase our low audio threshold, and to // increase it by more. @@ -814,20 +814,20 @@ private: TimeDuration decodeTime = TimeStamp::Now() - aDecodeStart; int64_t adjustedTime = THRESHOLD_FACTOR * DurationToUsecs(decodeTime); - if (adjustedTime > mMaster->mLowAudioThresholdUsecs.ToMicroseconds() + if (adjustedTime > mMaster->mLowAudioThreshold.ToMicroseconds() && !mMaster->HasLowBufferedData()) { - mMaster->mLowAudioThresholdUsecs = TimeUnit::FromMicroseconds( + mMaster->mLowAudioThreshold = TimeUnit::FromMicroseconds( std::min(adjustedTime, mMaster->mAmpleAudioThresholdUsecs)); mMaster->mAmpleAudioThresholdUsecs = - std::max(THRESHOLD_FACTOR * mMaster->mLowAudioThresholdUsecs.ToMicroseconds(), + std::max(THRESHOLD_FACTOR * mMaster->mLowAudioThreshold.ToMicroseconds(), mMaster->mAmpleAudioThresholdUsecs); SLOG("Slow video decode, set " "mLowAudioThresholdUsecs=%" PRId64 " mAmpleAudioThresholdUsecs=%" PRId64, - mMaster->mLowAudioThresholdUsecs.ToMicroseconds(), + mMaster->mLowAudioThreshold.ToMicroseconds(), mMaster->mAmpleAudioThresholdUsecs); } } @@ -2299,7 +2299,7 @@ DecodingState::NeedToSkipToNextKeyframe() !Reader()->IsAsync() && mMaster->IsAudioDecoding() && (mMaster->GetDecodedAudioDuration() - < mMaster->mLowAudioThresholdUsecs.ToMicroseconds() * mMaster->mPlaybackRate); + < mMaster->mLowAudioThreshold.ToMicroseconds() * mMaster->mPlaybackRate); bool isLowOnDecodedVideo = (mMaster->GetClock() - mMaster->mDecodedVideoEndTime) * mMaster->mPlaybackRate @@ -2594,7 +2594,7 @@ MediaDecoderStateMachine::MediaDecoderStateMachine(MediaDecoder* aDecoder, mDecodedAudioEndTime(0), mDecodedVideoEndTime(0), mPlaybackRate(1.0), - mLowAudioThresholdUsecs(detail::LOW_AUDIO_USECS), + mLowAudioThreshold(detail::LOW_AUDIO_THRESHOLD), mAmpleAudioThresholdUsecs(detail::AMPLE_AUDIO_USECS), mAudioCaptured(false), mMinimizePreroll(aDecoder->GetMinimizePreroll()), diff --git a/dom/media/MediaDecoderStateMachine.h b/dom/media/MediaDecoderStateMachine.h index 846c699c1b5b..39b76292203a 100644 --- a/dom/media/MediaDecoderStateMachine.h +++ b/dom/media/MediaDecoderStateMachine.h @@ -583,11 +583,11 @@ private: // decode video frames, in order to reduce the chance of audio underruns. // Note that we don't ever reset this threshold, it only ever grows as // we detect that the decode can't keep up with rendering. - media::TimeUnit mLowAudioThresholdUsecs; + media::TimeUnit mLowAudioThreshold; // Our "ample" audio threshold. Once we've this much audio decoded, we - // pause decoding. If we increase mLowAudioThresholdUsecs, we'll also - // increase this too appropriately (we don't want mLowAudioThresholdUsecs + // pause decoding. If we increase mLowAudioThreshold, we'll also + // increase this too appropriately (we don't want mLowAudioThreshold // to be greater than ampleAudioThreshold, else we'd stop decoding!). // Note that we don't ever reset this threshold, it only ever grows as // we detect that the decode can't keep up with rendering.