Source/WebCore/platform/audio/AudioBus.h

/*
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#ifndef AudioBus_h
#define AudioBus_h

#include "AudioChannel.h"
#include <memory>
#include <wtf/Noncopyable.h>
#include <wtf/ThreadSafeRefCounted.h>
#include <wtf/Vector.h>

namespace WebCore {

// An AudioBus represents a collection of one or more AudioChannels.
// The data layout is "planar" as opposed to "interleaved".
// An AudioBus with one channel is mono, an AudioBus with two channels is stereo, etc.
class AudioBus final : public ThreadSafeRefCounted<AudioBus> {
    WTF_MAKE_NONCOPYABLE(AudioBus);
public:
    enum {
        ChannelLeft = 0,
        ChannelRight = 1,
        ChannelCenter = 2, // center and mono are the same
        ChannelMono = 2,
        ChannelLFE = 3,
        ChannelSurroundLeft = 4,
        ChannelSurroundRight = 5,
    };

    enum {
        LayoutCanonical = 0
        // Can define non-standard layouts here
    };

    enum ChannelInterpretation {
        Speakers,
        Discrete,
    };

    // allocate indicates whether or not to initially have the AudioChannels created with managed storage.
    // Normal usage is to pass true here, in which case the AudioChannels will memory-manage their own storage.
    // If allocate is false then setChannelMemory() has to be called later on for each channel before the AudioBus is useable...
    static RefPtr<AudioBus> create(unsigned numberOfChannels, size_t length, bool allocate = true);

    // Tells the given channel to use an externally allocated buffer.
    void setChannelMemory(unsigned channelIndex, float* storage, size_t length);

    // Channels
    unsigned numberOfChannels() const { return m_channels.size(); }

    AudioChannel* channel(unsigned channel) { return m_channels[channel].get(); }
    const AudioChannel* channel(unsigned channel) const { return m_channels[channel].get(); }
    AudioChannel* channelByType(unsigned type);
    const AudioChannel* channelByType(unsigned type) const;

    // Number of sample-frames
    size_t length() const { return m_length; }

    // resizeSmaller() can only be called with a new length <= the current length.
    // The data stored in the bus will remain undisturbed.
    void resizeSmaller(size_t newLength);

    // Sample-rate : 0.0 if unknown or "don't care"
    float sampleRate() const { return m_sampleRate; }
    void setSampleRate(float sampleRate) { m_sampleRate = sampleRate; }

    // Zeroes all channels.
    void zero();

    // Clears the silent flag on all channels.
    void clearSilentFlag();

    // Returns true if the silent bit is set on all channels.
    bool isSilent() const;

    // Returns true if the channel count and frame-size match.
    bool topologyMatches(const AudioBus &sourceBus) const;

    // Creates a new buffer from a range in the source buffer.
    // 0 may be returned if the range does not fit in the sourceBuffer
    static RefPtr<AudioBus> createBufferFromRange(const AudioBus* sourceBuffer, unsigned startFrame, unsigned endFrame);


    // Creates a new AudioBus by sample-rate converting sourceBus to the newSampleRate.
    // setSampleRate() must have been previously called on sourceBus.
    // Note: sample-rate conversion is already handled in the file-reading code for the mac port, so we don't need this.
    static RefPtr<AudioBus> createBySampleRateConverting(const AudioBus* sourceBus, bool mixToMono, double newSampleRate);

    // Creates a new AudioBus by mixing all the channels down to mono.
    // If sourceBus is already mono, then the returned AudioBus will simply be a copy.
    static RefPtr<AudioBus> createByMixingToMono(const AudioBus* sourceBus);

    // Scales all samples by the same amount.
    void scale(float scale);

    void reset() { m_isFirstTime = true; } // for de-zippering

    // Copies the samples from the source bus to this one.
    void copyFromRange(const AudioBus& sourceBus, unsigned startFrame, unsigned endFrame);

    // Copies the samples from the source bus to this one.
    // This is just a simple per-channel copy if the number of channels match, otherwise an up-mix or down-mix is done.
    void copyFrom(const AudioBus& sourceBus, ChannelInterpretation = Speakers);

    // Sums the samples from the source bus to this one.
    // This is just a simple per-channel summing if the number of channels match, otherwise an up-mix or down-mix is done.
    void sumFrom(const AudioBus& sourceBus, ChannelInterpretation = Speakers);

    // Copy each channel from sourceBus into our corresponding channel.
    // We scale by targetGain (and our own internal gain m_busGain), performing "de-zippering" to smoothly change from *lastMixGain to (targetGain*m_busGain).
    // The caller is responsible for setting up lastMixGain to point to storage which is unique for every "stream" which will be applied to this bus.
    // This represents the dezippering memory.
    void copyWithGainFrom(const AudioBus &sourceBus, float* lastMixGain, float targetGain);

    // Copies the sourceBus by scaling with sample-accurate gain values.
    void copyWithSampleAccurateGainValuesFrom(const AudioBus &sourceBus, float* gainValues, unsigned numberOfGainValues);

    // Returns maximum absolute value across all channels (useful for normalization).
    float maxAbsValue() const;

    // Makes maximum absolute value == 1.0 (if possible).
    void normalize();

    static RefPtr<AudioBus> loadPlatformResource(const char* name, float sampleRate);

protected:
    AudioBus() { }

    AudioBus(unsigned numberOfChannels, size_t length, bool allocate);

    void speakersCopyFrom(const AudioBus&);
    void discreteCopyFrom(const AudioBus&);
    void speakersSumFrom(const AudioBus&);
    void discreteSumFrom(const AudioBus&);
    void speakersSumFrom5_1_ToMono(const AudioBus&);

    size_t m_length;
    Vector<std::unique_ptr<AudioChannel>> m_channels;
    int m_layout;
    float m_busGain;
    std::unique_ptr<AudioFloatArray> m_dezipperGainValues;
    bool m_isFirstTime;
    float m_sampleRate; // 0.0 if unknown or N/A
};

} // WebCore

#endif // AudioBus_h