| /* |
| * Copyright (C) 2010 Google Inc. All rights reserved. |
| * |
| * Redistribution and use in source and binary forms, with or without |
| * modification, are permitted provided that the following conditions |
| * are met: |
| * |
| * 1. Redistributions of source code must retain the above copyright |
| * notice, this list of conditions and the following disclaimer. |
| * 2. Redistributions in binary form must reproduce the above copyright |
| * notice, this list of conditions and the following disclaimer in the |
| * documentation and/or other materials provided with the distribution. |
| * 3. Neither the name of Apple Inc. ("Apple") nor the names of |
| * its contributors may be used to endorse or promote products derived |
| * from this software without specific prior written permission. |
| * |
| * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY |
| * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED |
| * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE |
| * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY |
| * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES |
| * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; |
| * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND |
| * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT |
| * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF |
| * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. |
| */ |
| |
| #pragma once |
| |
| #include "AudioContext.h" |
| #include "AudioParamTimeline.h" |
| #include "AudioSummingJunction.h" |
| #include <JavaScriptCore/Float32Array.h> |
| #include <sys/types.h> |
| #include <wtf/LoggerHelper.h> |
| #include <wtf/RefCounted.h> |
| #include <wtf/text/WTFString.h> |
| |
| namespace WebCore { |
| |
| class AudioNodeOutput; |
| |
| class AudioParam final |
| : public AudioSummingJunction |
| , public RefCounted<AudioParam> |
| #if !RELEASE_LOG_DISABLED |
| , private LoggerHelper |
| #endif |
| { |
| public: |
| static const double DefaultSmoothingConstant; |
| static const double SnapThreshold; |
| |
| static Ref<AudioParam> create(AudioContext& context, const String& name, double defaultValue, double minValue, double maxValue, unsigned units = 0) |
| { |
| return adoptRef(*new AudioParam(context, name, defaultValue, minValue, maxValue, units)); |
| } |
| |
| // AudioSummingJunction |
| bool canUpdateState() override { return true; } |
| void didUpdate() override { } |
| |
| // Intrinsic value. |
| float value(); |
| void setValue(float); |
| |
| // Final value for k-rate parameters, otherwise use calculateSampleAccurateValues() for a-rate. |
| // Must be called in the audio thread. |
| float finalValue(); |
| |
| String name() const { return m_name; } |
| |
| float minValue() const { return static_cast<float>(m_minValue); } |
| float maxValue() const { return static_cast<float>(m_maxValue); } |
| float defaultValue() const { return static_cast<float>(m_defaultValue); } |
| unsigned units() const { return m_units; } |
| |
| // Value smoothing: |
| |
| // When a new value is set with setValue(), in our internal use of the parameter we don't immediately jump to it. |
| // Instead we smoothly approach this value to avoid glitching. |
| float smoothedValue(); |
| |
| // Smoothly exponentially approaches to (de-zippers) the desired value. |
| // Returns true if smoothed value has already snapped exactly to value. |
| bool smooth(); |
| |
| void resetSmoothedValue() { m_smoothedValue = m_value; } |
| void setSmoothingConstant(double k) { m_smoothingConstant = k; } |
| |
| // Parameter automation. |
| void setValueAtTime(float value, float time) { m_timeline.setValueAtTime(value, time); } |
| void linearRampToValueAtTime(float value, float time) { m_timeline.linearRampToValueAtTime(value, time); } |
| void exponentialRampToValueAtTime(float value, float time) { m_timeline.exponentialRampToValueAtTime(value, time); } |
| void setTargetAtTime(float target, float time, float timeConstant) { m_timeline.setTargetAtTime(target, time, timeConstant); } |
| void setValueCurveAtTime(const RefPtr<Float32Array>& curve, float time, float duration) { m_timeline.setValueCurveAtTime(curve.get(), time, duration); } |
| void cancelScheduledValues(float startTime) { m_timeline.cancelScheduledValues(startTime); } |
| |
| bool hasSampleAccurateValues() { return m_timeline.hasValues() || numberOfRenderingConnections(); } |
| |
| // Calculates numberOfValues parameter values starting at the context's current time. |
| // Must be called in the context's render thread. |
| void calculateSampleAccurateValues(float* values, unsigned numberOfValues); |
| |
| // Connect an audio-rate signal to control this parameter. |
| void connect(AudioNodeOutput*); |
| void disconnect(AudioNodeOutput*); |
| |
| protected: |
| AudioParam(AudioContext&, const String&, double defaultValue, double minValue, double maxValue, unsigned units = 0); |
| |
| private: |
| // sampleAccurate corresponds to a-rate (audio rate) vs. k-rate in the Web Audio specification. |
| void calculateFinalValues(float* values, unsigned numberOfValues, bool sampleAccurate); |
| void calculateTimelineValues(float* values, unsigned numberOfValues); |
| |
| #if !RELEASE_LOG_DISABLED |
| const Logger& logger() const final { return m_logger.get(); } |
| const void* logIdentifier() const final { return m_logIdentifier; } |
| const char* logClassName() const final { return "AudioParam"; } |
| WTFLogChannel& logChannel() const final; |
| #endif |
| |
| String m_name; |
| double m_value; |
| double m_defaultValue; |
| double m_minValue; |
| double m_maxValue; |
| unsigned m_units; |
| |
| // Smoothing (de-zippering) |
| double m_smoothedValue; |
| double m_smoothingConstant; |
| |
| AudioParamTimeline m_timeline; |
| |
| #if !RELEASE_LOG_DISABLED |
| mutable Ref<const Logger> m_logger; |
| const void* m_logIdentifier; |
| #endif |
| }; |
| |
| } // namespace WebCore |