Mike Brady
fd880056fb
Add three parameters to the backend play() function call -- (1) a flag indicating whether the samples are timed or not. If timed, (2) the timestamp and (3) the local time at which the first frame should be heard.
Update missing pipewire library message.
Makefile.am fix.
Update the SHM version and fix the SHM name so that it works with FreeBSD.
Remove redundant (?) AC_HEADER_STDC check.
Fix compilation and installation under FreeBSD -- changes to allow building in a separate directory broke the FreeBSD build process.
Add AC_CHECK_LIB for gcrypt in case PKG_CHECK_MODULES fails to find it.
Find gcrypt using pkg-config
Add code to detect when the RTSP channel goes idle for a period, and use it to reset clocks and timings, etc. so that SPS resumes correctly where the source has gone to sleep while playing a realtime stream and has subsequently woken up.
Don't exit if a UDP Clock Control packet is empty. Also check minimum packet size.
Remove some very experimental code which may cause memory and port leaks.
Strip file path from the filename used in debug messages, information messages, warnings and fatal error messages.
BB fix to Makefile.am -- overwriting the DBUS stuff with MPRIS definitions, duh.
Fix some problems building on FreeBSD and tidy up the use of the "sed" editor.
Fix some problems building on picore.
Fix an uninitialised variable
start using the "nqptp" SMI
76 lines
2.5 KiB
C
76 lines
2.5 KiB
C
#ifndef _AUDIO_H
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#define _AUDIO_H
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#include <libconfig.h>
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#include <stdint.h>
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// clang-format off
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// Play samples provided may be from the source, in which case they will be timed
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// or they may be generated by Shairport Sync, in which case they will not be timed.
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// Typically these would be samples of silence, which may be dithered, sent during the lead-in to
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// the start of the material, or inserted instead of a missing frame, or after a flush.
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// clang-format on
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typedef enum {
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play_samples_are_untimed = 0, // typically the samples are (possibly dithered) silence
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play_samples_are_timed, // timed and numbered.
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} play_samples_type;
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typedef struct {
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double current_volume_dB;
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int32_t minimum_volume_dB;
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int32_t maximum_volume_dB;
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} audio_parameters;
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typedef struct {
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void (*help)(void);
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char *name;
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// start of program
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int (*init)(int argc, char **argv);
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// at end of program
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void (*deinit)(void);
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int (*prepare)(void); // looks and sets stuff in the config data structure
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void (*start)(int sample_rate, int sample_format);
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// block of samples
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int (*play)(void *buf, int samples, int sample_type, uint32_t timestamp, uint64_t playtime);
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void (*stop)(void);
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// may be null if no implemented
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int (*is_running)(
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void); // if implemented, will return 0 if everything is okay, non-zero otherwise
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// may be null if not implemented
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void (*flush)(void);
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// returns the delay before the next frame to be sent to the device would actually be audible.
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// almost certainly wrong if the buffer is empty, so put silent buffers into it to make it busy.
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// will change dynamically, so keep watching it. Implemented in ALSA only.
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// returns a negative error code if there's a problem
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int (*delay)(long *the_delay); // snd_pcm_sframes_t is a signed long
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int (*stats)(uint64_t *raw_measurement_time, uint64_t *corrected_measurement_time,
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uint64_t *delay,
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uint64_t *frames_sent_to_dac); // use this to get the true rate of the DAC
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// may be NULL, in which case soft volume is applied
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void (*volume)(double vol);
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// may be NULL, in which case soft volume parameters are used
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void (*parameters)(audio_parameters *info);
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// may be NULL, in which case software muting is used.
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// also, will return a 1 if it is actually using the mute facility, 0 otherwise
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int (*mute)(int do_mute);
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} audio_output;
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audio_output *audio_get_output(const char *name);
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void audio_ls_outputs(void);
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void parse_general_audio_options(void);
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#endif //_AUDIO_H
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