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https://github.com/OpenMW/openmw.git
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20e3eeddde
2a68d945cdadded const version of a callback functions but didn't enable them. They were guarded by a version check:2a68d945cd/libavformat/version_major.h (L48)So for anything LIBAVFORMAT_VERSION_MAJOR < 61 they are not enabled therefore they are enabled for everything >= 61.0.100. See https://github.com/elsid/openmw/actions/runs/10255993574/job/28374152796 as example of failure when building with 60.16.100.
122 lines
3.3 KiB
C++
122 lines
3.3 KiB
C++
#ifndef VIDEOPLAYER_AUDIODECODER_H
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#define VIDEOPLAYER_AUDIODECODER_H
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#include <stdint.h>
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#include <new>
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#include <memory>
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#include <extern/osg-ffmpeg-videoplayer/libavutildefines.hpp>
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#if defined(_MSC_VER)
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#pragma warning (push)
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#pragma warning (disable : 4244)
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#endif
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extern "C"
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{
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#include <libavutil/avutil.h>
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#include <libavcodec/avcodec.h>
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#include <libavformat/avformat.h>
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#include <libavutil/channel_layout.h>
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}
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#if defined(_MSC_VER)
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#pragma warning (pop)
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#endif
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#if defined(_WIN32) && !defined(__MINGW32__)
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#include <basetsd.h>
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typedef SSIZE_T ssize_t;
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#endif
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namespace Video
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{
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struct AudioResampler;
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struct VideoState;
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class MovieAudioDecoder
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{
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protected:
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VideoState *mVideoState;
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AVCodecContext* mAudioContext;
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AVStream *mAVStream;
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enum AVSampleFormat mOutputSampleFormat;
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#if OPENMW_FFMPEG_5_OR_GREATER
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AVChannelLayout mOutputChannelLayout;
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#else
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uint64_t mOutputChannelLayout;
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#endif
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int mOutputSampleRate;
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ssize_t mFramePos;
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ssize_t mFrameSize;
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double mAudioClock;
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private:
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struct AutoAVPacket : public AVPacket {
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AutoAVPacket(int size=0)
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{
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if(av_new_packet(this, size) < 0)
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throw std::bad_alloc();
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}
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~AutoAVPacket()
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{ av_packet_unref(this); }
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};
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std::unique_ptr<AudioResampler> mAudioResampler;
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uint8_t *mDataBuf;
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uint8_t **mFrameData;
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int mDataBufLen;
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AutoAVPacket mPacket;
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AVFrame *mFrame;
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bool mGetNextPacket;
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/* averaging filter for audio sync */
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double mAudioDiffAccum;
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const double mAudioDiffAvgCoef;
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const double mAudioDiffThreshold;
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int mAudioDiffAvgCount;
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/* Add or subtract samples to get a better sync, return number of bytes to
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* skip (negative means to duplicate). */
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int synchronize_audio();
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/// @param sample_skip If seeking happened, the sample_skip variable will be reset to 0.
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int audio_decode_frame(AVFrame *frame, int &sample_skip);
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public:
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MovieAudioDecoder(VideoState *is);
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virtual ~MovieAudioDecoder();
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int getOutputSampleRate() const;
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AVSampleFormat getOutputSampleFormat() const;
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uint64_t getOutputChannelLayout() const;
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void setupFormat();
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/// Adjust the given audio settings to the application's needs. The data given by the read() function will
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/// be in the desired format written to this function's parameters.
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/// @par Depending on the application, we may want either fixed settings, or a "closest supported match"
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/// for the input that does not incur precision loss (e.g. planar -> non-planar format).
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virtual void adjustAudioSettings(AVSampleFormat& sampleFormat, uint64_t& channelLayout, int& sampleRate) = 0;
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/// Return the current offset in seconds from the beginning of the audio stream.
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/// @par An internal clock exists in the mAudioClock member, and is used in the default implementation. However,
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/// for an accurate clock, it's best to also take the current offset in the audio buffer into account.
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virtual double getAudioClock();
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/// This is the main interface to be used by the user's audio library.
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/// @par Request filling the \a stream with \a len number of bytes.
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/// @return The number of bytes read (may not be the requested number if we arrived at the end of the audio stream)
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size_t read(char *stream, size_t len);
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};
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}
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#endif
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