Stability improvements -- take more care when the fifi is full, fix a logical error...

[thirdparty/shairport-sync.git] / audio.h

#ifndef _AUDIO_H
#define _AUDIO_H

#include <libconfig.h>
#include <stdint.h>

// clang-format off
// Play samples provided may be from the source, in which case they will be timed
// or they may be generated by Shairport Sync, in which case they will not be timed.

// Typically these would be samples of silence, which may be dithered, sent during the lead-in to 
// the start of the material, or inserted instead of a missing frame, or after a flush.
// clang-format on

typedef enum {
  play_samples_are_untimed = 0, // typically the samples are (possibly dithered) silence
  play_samples_are_timed,       // timed and numbered.
} play_samples_type;

typedef struct {
  double current_volume_dB;
  int32_t minimum_volume_dB;
  int32_t maximum_volume_dB;
} audio_parameters;

typedef struct {
  void (*help)(void);
  char *name;

  // start of program
  int (*init)(int argc, char **argv);
  // at end of program
  void (*deinit)(void);

  int (*prepare)(void); // looks and sets stuff in the config data structure

  void (*start)(int sample_rate, int sample_format);

  // block of samples
  int (*play)(void *buf, int samples, int sample_type, uint32_t timestamp, uint64_t playtime);
  void (*stop)(void);

  // may be null if no implemented
  int (*is_running)(
      void); // if implemented, will return 0 if everything is okay, non-zero otherwise

  // may be null if not implemented
  void (*flush)(void);

  // returns the delay before the next frame to be sent to the device would actually be audible.
  // almost certainly wrong if the buffer is empty, so put silent buffers into it to make it busy.
  // will change dynamically, so keep watching it. Implemented in ALSA only.
  // returns a negative error code if there's a problem
  int (*delay)(long *the_delay); // snd_pcm_sframes_t is a signed long
  int (*stats)(uint64_t *raw_measurement_time, uint64_t *corrected_measurement_time,
               uint64_t *delay,
               uint64_t *frames_sent_to_dac); // use this to get the true rate of the DAC

  // may be NULL, in which case soft volume is applied
  void (*volume)(double vol);

  // may be NULL, in which case soft volume parameters are used
  void (*parameters)(audio_parameters *info);

  // may be NULL, in which case software muting is used.
  // also, will return a 1 if it is actually using the mute facility, 0 otherwise
  int (*mute)(int do_mute);

} audio_output;

audio_output *audio_get_output(const char *name);
void audio_ls_outputs(void);
void parse_general_audio_options(void);

#endif //_AUDIO_H