/* * 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. * * THIS SOFTWARE IS PROVIDED BY APPLE INC. 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 INC. 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 "AudioScheduledSourceNode.h" #include <wtf/Lock.h> namespace WebCore { class AudioBuffer; class PannerNode; // AudioBufferSourceNode is an AudioNode representing an audio source from an in-memory audio asset represented by an AudioBuffer. // It generally will be used for short sounds which require a high degree of scheduling flexibility (can playback in rhythmically perfect ways). class AudioBufferSourceNode final : public AudioScheduledSourceNode { public: static Ref<AudioBufferSourceNode> create(AudioContext&, float sampleRate); virtual ~AudioBufferSourceNode(); // AudioNode void process(size_t framesToProcess) final; void reset() final; // setBuffer() is called on the main thread. This is the buffer we use for playback. // returns true on success. void setBuffer(RefPtr<AudioBuffer>&&); AudioBuffer* buffer() { return m_buffer.get(); } // numberOfChannels() returns the number of output channels. This value equals the number of channels from the buffer. // If a new buffer is set with a different number of channels, then this value will dynamically change. unsigned numberOfChannels(); // Play-state ExceptionOr<void> start(double when, double grainOffset, std::optional<double> grainDuration); // Note: the attribute was originally exposed as .looping, but to be more consistent in naming with <audio> // and with how it's described in the specification, the proper attribute name is .loop // The old attribute is kept for backwards compatibility. bool loop() const { return m_isLooping; } void setLoop(bool looping) { m_isLooping = looping; } // Loop times in seconds. double loopStart() const { return m_loopStart; } double loopEnd() const { return m_loopEnd; } void setLoopStart(double loopStart) { m_loopStart = loopStart; } void setLoopEnd(double loopEnd) { m_loopEnd = loopEnd; } // Deprecated. bool looping(); void setLooping(bool); AudioParam* gain() { return m_gain.get(); } AudioParam* playbackRate() { return m_playbackRate.get(); } // If a panner node is set, then we can incorporate doppler shift into the playback pitch rate. void setPannerNode(PannerNode*); void clearPannerNode(); // If we are no longer playing, propogate silence ahead to downstream nodes. bool propagatesSilence() const final; // AudioScheduledSourceNode void finish() final; private: AudioBufferSourceNode(AudioContext&, float sampleRate); double tailTime() const final { return 0; } double latencyTime() const final { return 0; } enum BufferPlaybackMode { Entire, Partial }; ExceptionOr<void> startPlaying(BufferPlaybackMode, double when, double grainOffset, double grainDuration); // Returns true on success. bool renderFromBuffer(AudioBus*, unsigned destinationFrameOffset, size_t numberOfFrames); // Render silence starting from "index" frame in AudioBus. inline bool renderSilenceAndFinishIfNotLooping(AudioBus*, unsigned index, size_t framesToProcess); // m_buffer holds the sample data which this node outputs. RefPtr<AudioBuffer> m_buffer; // Pointers for the buffer and destination. std::unique_ptr<const float*[]> m_sourceChannels; std::unique_ptr<float*[]> m_destinationChannels; // Used for the "gain" and "playbackRate" attributes. RefPtr<AudioParam> m_gain; RefPtr<AudioParam> m_playbackRate; // If m_isLooping is false, then this node will be done playing and become inactive after it reaches the end of the sample data in the buffer. // If true, it will wrap around to the start of the buffer each time it reaches the end. bool m_isLooping; double m_loopStart; double m_loopEnd; // m_virtualReadIndex is a sample-frame index into our buffer representing the current playback position. // Since it's floating-point, it has sub-sample accuracy. double m_virtualReadIndex; // Granular playback bool m_isGrain; double m_grainOffset; // in seconds double m_grainDuration; // in seconds // totalPitchRate() returns the instantaneous pitch rate (non-time preserving). // It incorporates the base pitch rate, any sample-rate conversion factor from the buffer, and any doppler shift from an associated panner node. double totalPitchRate(); // m_lastGain provides continuity when we dynamically adjust the gain. float m_lastGain; // We optionally keep track of a panner node which has a doppler shift that is incorporated into // the pitch rate. We manually manage ref-counting because we want to use RefTypeConnection. PannerNode* m_pannerNode; // This synchronizes process() with setBuffer() which can cause dynamic channel count changes. mutable Lock m_processMutex; }; } // namespace WebCore