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#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
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