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Side by Side Diff: third_party/WebKit/Source/modules/webaudio/AudioBufferSourceNode.cpp

Issue 2501863003: Support for AudioContextOptions latencyHint. (Closed)
Patch Set: Fixes to WebAudioDeviceImpl unit test. Created 3 years, 10 months ago
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1 /* 1 /*
2 * Copyright (C) 2010, Google Inc. All rights reserved. 2 * Copyright (C) 2010, Google Inc. All rights reserved.
3 * 3 *
4 * Redistribution and use in source and binary forms, with or without 4 * Redistribution and use in source and binary forms, with or without
5 * modification, are permitted provided that the following conditions 5 * modification, are permitted provided that the following conditions
6 * are met: 6 * are met:
7 * 1. Redistributions of source code must retain the above copyright 7 * 1. Redistributions of source code must retain the above copyright
8 * notice, this list of conditions and the following disclaimer. 8 * notice, this list of conditions and the following disclaimer.
9 * 2. Redistributions in binary form must reproduce the above copyright 9 * 2. Redistributions in binary form must reproduce the above copyright
10 * notice, this list of conditions and the following disclaimer in the 10 * notice, this list of conditions and the following disclaimer in the
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555 } 555 }
556 556
557 double AudioBufferSourceHandler::computePlaybackRate() { 557 double AudioBufferSourceHandler::computePlaybackRate() {
558 // Incorporate buffer's sample-rate versus BaseAudioContext's sample-rate. 558 // Incorporate buffer's sample-rate versus BaseAudioContext's sample-rate.
559 // Normally it's not an issue because buffers are loaded at the 559 // Normally it's not an issue because buffers are loaded at the
560 // BaseAudioContext's sample-rate, but we can handle it in any case. 560 // BaseAudioContext's sample-rate, but we can handle it in any case.
561 double sampleRateFactor = 1.0; 561 double sampleRateFactor = 1.0;
562 if (buffer()) { 562 if (buffer()) {
563 // Use doubles to compute this to full accuracy. 563 // Use doubles to compute this to full accuracy.
564 sampleRateFactor = 564 sampleRateFactor =
565 buffer()->sampleRate() / static_cast<double>(sampleRate()); 565 buffer()->sampleRate() / static_cast<double>(context()->sampleRate());
566 } 566 }
567 567
568 // Use finalValue() to incorporate changes of AudioParamTimeline and 568 // Use finalValue() to incorporate changes of AudioParamTimeline and
569 // AudioSummingJunction from m_playbackRate AudioParam. 569 // AudioSummingJunction from m_playbackRate AudioParam.
570 double basePlaybackRate = m_playbackRate->finalValue(); 570 double basePlaybackRate = m_playbackRate->finalValue();
571 571
572 double finalPlaybackRate = sampleRateFactor * basePlaybackRate; 572 double finalPlaybackRate = sampleRateFactor * basePlaybackRate;
573 573
574 // Take the detune value into account for the final playback rate. 574 // Take the detune value into account for the final playback rate.
575 finalPlaybackRate *= pow(2, m_detune->finalValue() / 1200); 575 finalPlaybackRate *= pow(2, m_detune->finalValue() / 1200);
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755 755
756 void AudioBufferSourceNode::start(double when, 756 void AudioBufferSourceNode::start(double when,
757 double grainOffset, 757 double grainOffset,
758 double grainDuration, 758 double grainDuration,
759 ExceptionState& exceptionState) { 759 ExceptionState& exceptionState) {
760 audioBufferSourceHandler().start(when, grainOffset, grainDuration, 760 audioBufferSourceHandler().start(when, grainOffset, grainDuration,
761 exceptionState); 761 exceptionState);
762 } 762 }
763 763
764 } // namespace blink 764 } // namespace blink
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