#ifndef VIDEOPLAYER_AUDIODECODER_H
#define VIDEOPLAYER_AUDIODECODER_H

#include <stdint.h>

#include <new>
#include <memory>

#include <extern/osg-ffmpeg-videoplayer/libavutildefines.hpp>

#if defined(_MSC_VER)
    #pragma warning (push)
    #pragma warning (disable : 4244)
#endif

extern "C"
{
    #include <libavutil/avutil.h>
    #include <libavcodec/avcodec.h>
    #include <libavformat/avformat.h>
    #include <libavutil/channel_layout.h>
}

#if defined(_MSC_VER)
    #pragma warning (pop)
#endif

#if defined(_WIN32) && !defined(__MINGW32__)
#include <basetsd.h>

typedef SSIZE_T ssize_t;
#endif

namespace Video
{

struct AudioResampler;

struct VideoState;

class MovieAudioDecoder
{
protected:
    VideoState *mVideoState;
    AVCodecContext* mAudioContext;
    AVStream *mAVStream;
    enum AVSampleFormat mOutputSampleFormat;
    #if OPENMW_FFMPEG_5_OR_GREATER
    AVChannelLayout mOutputChannelLayout;
    #else
    uint64_t mOutputChannelLayout;
    #endif
    int mOutputSampleRate;
    ssize_t mFramePos;
    ssize_t mFrameSize;
    double mAudioClock;

private:
    struct AutoAVPacket : public AVPacket {
        AutoAVPacket(int size=0)
        {
            if(av_new_packet(this, size) < 0)
                throw std::bad_alloc();
        }
        ~AutoAVPacket()
        { av_packet_unref(this); }
    };


    std::unique_ptr<AudioResampler> mAudioResampler;

    uint8_t *mDataBuf;
    uint8_t **mFrameData;
    int mDataBufLen;

    AutoAVPacket mPacket;
    AVFrame *mFrame;
    bool mGetNextPacket;

    /* averaging filter for audio sync */
    double mAudioDiffAccum;
    const double mAudioDiffAvgCoef;
    const double mAudioDiffThreshold;
    int mAudioDiffAvgCount;

    /* Add or subtract samples to get a better sync, return number of bytes to
     * skip (negative means to duplicate). */
    int synchronize_audio();

    /// @param sample_skip If seeking happened, the sample_skip variable will be reset to 0.
    int audio_decode_frame(AVFrame *frame, int &sample_skip);

public:
    MovieAudioDecoder(VideoState *is);
    virtual ~MovieAudioDecoder();

    int getOutputSampleRate() const;
    AVSampleFormat getOutputSampleFormat() const;
    uint64_t getOutputChannelLayout() const;

    void setupFormat();

    /// Adjust the given audio settings to the application's needs. The data given by the read() function will
    /// be in the desired format written to this function's parameters.
    /// @par Depending on the application, we may want either fixed settings, or a "closest supported match"
    /// for the input that does not incur precision loss (e.g. planar -> non-planar format).
    virtual void adjustAudioSettings(AVSampleFormat& sampleFormat, uint64_t& channelLayout, int& sampleRate) = 0;

    /// Return the current offset in seconds from the beginning of the audio stream.
    /// @par An internal clock exists in the mAudioClock member, and is used in the default implementation. However,
    /// for an accurate clock, it's best to also take the current offset in the audio buffer into account.
    virtual double getAudioClock();

    /// This is the main interface to be used by the user's audio library.
    /// @par Request filling the \a stream with \a len number of bytes.
    /// @return The number of bytes read (may not be the requested number if we arrived at the end of the audio stream)
    size_t read(char *stream, size_t len);
};

}

#endif
