зеркало из https://github.com/mozilla/gecko-dev.git
Bug 869224 - Use the same algorithm to compute the buffer start/end time and to convert AudioParam time values to ticks; r=roc
This commit is contained in:
Ehsan Akhgari
Родитель
5db083ef7f
Коммит
f2d6931593
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@ -53,10 +53,9 @@ void
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AudioNodeStream::SetStreamTimeParameterImpl(uint32_t aIndex, MediaStream* aRelativeToStream,
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double aStreamTime)
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{
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StreamTime streamTime = std::max<MediaTime>(0, SecondsToMediaTime(aStreamTime));
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GraphTime graphTime = aRelativeToStream->StreamTimeToGraphTime(streamTime);
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StreamTime thisStreamTime = GraphTimeToStreamTimeOptimistic(graphTime);
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TrackTicks ticks = TimeToTicksRoundUp(IdealAudioRate(), thisStreamTime);
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TrackTicks ticks =
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WebAudioUtils::ConvertDestinationStreamTimeToSourceStreamTime(
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aStreamTime, this, aRelativeToStream);
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mEngine->SetStreamTimeParameter(aIndex, ticks);
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}
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@ -18,22 +18,29 @@ struct ConvertTimeToTickHelper
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static int64_t Convert(double aTime, void* aClosure)
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{
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TrackRate sampleRate = IdealAudioRate();
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StreamTime streamTime;
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ConvertTimeToTickHelper* This = static_cast<ConvertTimeToTickHelper*> (aClosure);
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if (This->mSourceStream) {
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TrackTicks tick = This->mDestinationStream->GetCurrentPosition();
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StreamTime destinationStreamTime = TicksToTimeRoundDown(sampleRate, tick);
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GraphTime graphTime = This->mDestinationStream->StreamTimeToGraphTime(destinationStreamTime);
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streamTime = This->mSourceStream->GraphTimeToStreamTime(graphTime);
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return WebAudioUtils::ConvertDestinationStreamTimeToSourceStreamTime(
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aTime, This->mSourceStream, This->mDestinationStream);
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} else {
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streamTime = This->mDestinationStream->GetCurrentPosition();
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StreamTime streamTime = This->mDestinationStream->GetCurrentPosition();
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return TimeToTicksRoundUp(IdealAudioRate(), streamTime + SecondsToMediaTime(aTime));
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}
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return TimeToTicksRoundDown(sampleRate, streamTime + SecondsToMediaTime(aTime));
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}
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};
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TrackTicks
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WebAudioUtils::ConvertDestinationStreamTimeToSourceStreamTime(double aTime,
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MediaStream* aSource,
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MediaStream* aDestination)
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{
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StreamTime streamTime = std::max<MediaTime>(0, SecondsToMediaTime(aTime));
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GraphTime graphTime = aDestination->StreamTimeToGraphTime(streamTime);
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StreamTime thisStreamTime = aSource->GraphTimeToStreamTimeOptimistic(graphTime);
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TrackTicks ticks = TimeToTicksRoundUp(IdealAudioRate(), thisStreamTime);
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return ticks;
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}
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double
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WebAudioUtils::StreamPositionToDestinationTime(TrackTicks aSourcePosition,
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AudioNodeStream* aSource,
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@ -41,6 +41,15 @@ struct WebAudioUtils {
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return 1.0 - std::exp(-1.0 / (aDuration * aSampleRate));
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}
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/**
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* Convert a time in second relative to the destination stream to
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* TrackTicks relative to the source stream.
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*/
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static TrackTicks
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ConvertDestinationStreamTimeToSourceStreamTime(double aTime,
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MediaStream* aSource,
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MediaStream* aDestination);
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/**
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* Converts AudioParamTimeline floating point time values to tick values
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* with respect to a source and a destination AudioNodeStream.
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@ -31,6 +31,7 @@ MOCHITEST_FILES := \
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test_audioParamLinearRamp.html \
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test_audioParamSetCurveAtTime.html \
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test_audioParamSetTargetAtTime.html \
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test_audioParamTimelineDestinationOffset.html \
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test_audioBufferSourceNode.html \
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test_audioBufferSourceNodeLazyLoopParam.html \
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test_audioBufferSourceNodeLoop.html \
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@ -0,0 +1,55 @@
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<!DOCTYPE HTML>
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<html>
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<head>
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<title>Test AudioParam timeline events scheduled after the destination stream has started playback</title>
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<script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
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<script type="text/javascript" src="webaudio.js"></script>
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<link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
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</head>
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<body>
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<pre id="test">
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<script class="testbody" type="text/javascript">
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SimpleTest.waitForExplicitFinish();
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addLoadEvent(function() {
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SpecialPowers.setBoolPref("media.webaudio.enabled", true);
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var context = new AudioContext();
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var sourceBuffer = context.createBuffer(1, 2048, context.sampleRate);
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for (var i = 0; i < 2048; ++i) {
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sourceBuffer.getChannelData(0)[i] = 1;
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}
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var emptyBuffer = context.createBuffer(1, 16384, context.sampleRate);
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setTimeout(function() {
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var source = context.createBufferSource();
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source.buffer = sourceBuffer;
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source.start(context.currentTime);
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source.stop(context.currentTime + sourceBuffer.duration);
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var gain = context.createGain();
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gain.gain.setValueAtTime(0, context.currentTime);
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gain.gain.setTargetAtTime(0, context.currentTime + sourceBuffer.duration, 1);
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source.connect(gain);
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var sp = context.createScriptProcessor(16384, 1);
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gain.connect(sp);
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sp.connect(context.destination);
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sp.onaudioprocess = function(e) {
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is(e.inputBuffer.numberOfChannels, 1, "Correct input channel count");
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compareBuffers(e.inputBuffer.getChannelData(0), emptyBuffer.getChannelData(0));
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sp.onaudioprocess = null;
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SpecialPowers.clearUserPref("media.webaudio.enabled");
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SimpleTest.finish();
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};
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}, 100);
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});
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</script>
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</pre>
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</body>
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</html>
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