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winamp/Src/replicant/audio/ifc_audioout.h
2024-09-24 14:54:57 +02:00

77 lines
3.1 KiB
C++

#pragma once
#include "foundation/dispatch.h"
#include "foundation/error.h"
#include "audio/parameters.h"
class NOVTABLE ifc_audioout : public Wasabi2::Dispatchable
{
protected:
ifc_audioout() : Dispatchable(DISPATCHABLE_VERSION) {}
~ifc_audioout() {}
public:
enum
{
CHANNEL_LAYOUT_MICROSOFT = 0x0, // microsoft channel order - http://www.microsoft.com/whdc/device/audio/multichaud.mspx#E4C
CHANNEL_LAYOUT_MPEG = 0x1,
};
enum
{
EXTENDED_FLAG_APPLY_GAIN=0x1, /* apply the gain value specified in Parameters::gain */
EXTENDED_FLAG_REPLAYGAIN=0x2, /* pass if you tried to figure out ReplayGain on your own. otherwise the Audio Output object will apply the default gain */
EXTENDED_FLAG_GAIN_MASK=EXTENDED_FLAG_APPLY_GAIN|EXTENDED_FLAG_REPLAYGAIN, /* a mask to check whether or not the gain value is valid */
/* so that you can check if a flag was set that you don't understand */
EXTENDED_FLAG_VALID_MASK=EXTENDED_FLAG_APPLY_GAIN|EXTENDED_FLAG_REPLAYGAIN,
};
struct Parameters
{
size_t sizeof_parameters;
nsaudio::Parameters audio;
/* anything after this needs sizeof_parameters to be large enough
AND a flag set in extended_fields_flags
if there's no flag for the field, it's because a default value of 0 can be assumed */
unsigned int extended_fields_flags; // set these if you use any of the following fields. see comment above
double gain; // additional gain specified by client. usually used for replaygain (so it can be combined with EQ pre-amp or float/pcm conversion)
size_t frames_trim_start; // number of frames to trim from the start
size_t frames_trim_end; // number of frames to trim from the start
};
int Output(const void *data, size_t data_size) { return AudioOutput_Output(data, data_size); }
// returns number of bytes that you can write
size_t CanWrite() { return AudioOutput_CanWrite(); }
void Flush(double seconds) { AudioOutput_Flush(seconds); }
void Pause(int state) { AudioOutput_Pause(state); }
/* called by the input plugin when no more output will be sent */
void Done() { AudioOutput_Done(); }
/* called by the input plugin when playback was forcefully stopped */
void Stop() { AudioOutput_Stop(); }
/* returns the latency in seconds (how many seconds until samples you're about to write show up at the audio output */
double Latency() { return AudioOutput_Latency(); }
/* only valid after a call to Done(). Returns NErr_True if there is still data in the buffer, NErr_False otherwise */
int Playing() { return AudioOutput_Playing(); }
protected:
virtual int WASABICALL AudioOutput_Output(const void *data, size_t data_size)=0;
virtual size_t WASABICALL AudioOutput_CanWrite()=0; // returns number of bytes that you can write
virtual void WASABICALL AudioOutput_Flush(double seconds)=0;
virtual void WASABICALL AudioOutput_Pause(int state)=0;
/* called by the input plugin when no more output will be sent */
virtual void WASABICALL AudioOutput_Done()=0;
/* called by the input plugin when playback was forcefully stopped */
virtual void WASABICALL AudioOutput_Stop()=0;
virtual double WASABICALL AudioOutput_Latency()=0;
virtual int WASABICALL AudioOutput_Playing()=0;
enum
{
DISPATCHABLE_VERSION,
};
};