Initial community commit

Src/replicant/audio/ifc_audioout.h

@@ -0,0 +1,76 @@
+#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,
+	};
+};