#ifndef _AUDIO_H #define _AUDIO_H #include #include 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); 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 (*rate_info)(uint64_t *elapsed_time, uint64_t *frames_played); // 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(char *name); void audio_ls_outputs(void); void parse_general_audio_options(void); #endif //_AUDIO_H