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player.h
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player.h
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#ifndef _PLAYER_H
#define _PLAYER_H
#include <arpa/inet.h>
#include <pthread.h>
#include "config.h"
#include "definitions.h"
#ifdef CONFIG_MBEDTLS
#include <mbedtls/aes.h>
#include <mbedtls/havege.h>
#endif
#ifdef CONFIG_POLARSSL
#include <polarssl/aes.h>
#include <polarssl/havege.h>
#endif
#ifdef CONFIG_OPENSSL
#include <openssl/aes.h>
#endif
#ifdef CONFIG_AIRPLAY_2
#include "pair_ap/pair.h"
#include <plist/plist.h>
#endif
#include "alac.h"
#include "audio.h"
#define time_ping_history_power_of_two 7
// this must now be zero, otherwise bad things will happen
#define time_ping_history \
(1 << time_ping_history_power_of_two) // 2^7 is 128. At 1 per three seconds, approximately six
// minutes of records
typedef struct time_ping_record {
uint64_t dispersion;
uint64_t local_time;
uint64_t remote_time;
int sequence_number;
int chosen;
} time_ping_record;
// these are for reporting the status of the clock
typedef enum {
clock_no_anchor_info,
clock_ok,
clock_service_unavailable,
clock_access_error,
clock_data_unavailable,
clock_no_master,
clock_version_mismatch,
clock_not_synchronised,
clock_not_valid,
clock_not_ready,
} clock_status_t;
typedef uint16_t seq_t;
typedef struct audio_buffer_entry { // decoded audio packets
uint8_t ready;
uint8_t status; // flags
uint16_t resend_request_number;
signed short *data;
seq_t sequence_number;
uint64_t initialisation_time; // the time the packet was added or the time it was noticed the
// packet was missing
uint64_t resend_time; // time of last resend request or zero
uint32_t given_timestamp; // for debugging and checking
int length; // the length of the decoded data
} abuf_t;
typedef struct stats { // statistics for running averages
int64_t sync_error, correction, drift;
} stats_t;
// default buffer size
// This needs to be a power of 2 because of the way BUFIDX(seqno) works.
// 512 is the minimum for normal operation -- it gives 512*352/44100 or just over 4 seconds of
// buffers.
// For at least 10 seconds, you need to go to 2048.
// Resend requests will be spaced out evenly in the latency period, subject to a minimum interval of
// about 0.25 seconds.
// Each buffer occupies 352*4 bytes plus about, say, 64 bytes of overhead in various places, say
// roughly 1,500 bytes per buffer.
// Thus, 2048 buffers will occupy about 3 megabytes -- no big deal in a normal machine but maybe a
// problem in an embedded device.
#define BUFFER_FRAMES 1024
typedef enum {
ast_unknown,
ast_uncompressed, // L16/44100/2
ast_apple_lossless,
} audio_stream_type;
typedef struct {
int encrypted;
uint8_t aesiv[16], aeskey[16];
int32_t fmtp[12];
audio_stream_type type;
} stream_cfg;
// the following is used even when not built for AirPlay 2
typedef enum {
unspecified_stream_category = 0,
ptp_stream,
ntp_stream,
remote_control_stream
} airplay_stream_c; // "c" for category
#ifdef CONFIG_AIRPLAY_2
typedef enum { ts_ntp, ts_ptp } timing_t;
typedef enum { ap_1, ap_2 } airplay_t;
typedef enum { realtime_stream, buffered_stream } airplay_stream_t;
typedef struct {
uint8_t *data;
size_t length;
size_t size;
} sized_buffer;
typedef struct {
struct pair_cipher_context *cipher_ctx;
sized_buffer encrypted_read_buffer;
sized_buffer plaintext_read_buffer;
int is_encrypted;
} pair_cipher_bundle; // cipher context and buffers
typedef struct {
struct pair_setup_context *setup_ctx;
struct pair_verify_context *verify_ctx;
pair_cipher_bundle control_cipher_bundle;
} ap2_pairing;
// flush requests are stored in order of flushFromSeq
// on the basis that block numbers are monotonic modulo 2^24
typedef struct flush_request_t {
int flushNow; // if true, the flushFrom stuff is invalid
uint32_t flushFromSeq;
uint32_t flushFromTS;
uint32_t flushUntilSeq;
uint32_t flushUntilTS;
struct flush_request_t *next;
} flush_request_t;
#endif
typedef struct {
int connection_number; // for debug ID purposes, nothing else...
int resend_interval; // this is really just for debugging
int rtsp_link_is_idle; // if true, this indicates if the client asleep
char *UserAgent; // free this on teardown
int AirPlayVersion; // zero if not an AirPlay session. Used to help calculate latency
int latency_warning_issued;
uint32_t latency; // the actual latency used for this play session
uint32_t minimum_latency; // set if an a=min-latency: line appears in the ANNOUNCE message; zero
// otherwise
uint32_t maximum_latency; // set if an a=max-latency: line appears in the ANNOUNCE message; zero
// otherwise
int software_mute_enabled; // if we don't have a real mute that we can use
int fd;
int authorized; // set if a password is required and has been supplied
char *auth_nonce; // the session nonce, if needed
stream_cfg stream;
SOCKADDR remote, local;
volatile int stop;
volatile int running;
volatile uint64_t watchdog_bark_time;
volatile int watchdog_barks; // number of times the watchdog has timed out and done something
int unfixable_error_reported; // set when an unfixable error command has been executed.
uint64_t playstart;
uint64_t connection_start_time; // the time the device is selected, which could be a long time
// before a play
pthread_t thread, timer_requester, rtp_audio_thread, rtp_control_thread, rtp_timing_thread,
player_watchdog_thread;
// buffers to delete on exit
int32_t *tbuf;
int32_t *sbuf;
char *outbuf;
// for generating running statistics...
stats_t *statistics;
// for holding the output rate information until printed out at the end of a session
double raw_frame_rate;
double corrected_frame_rate;
int frame_rate_valid;
// for holding input rate information until printed out at the end of a session
double input_frame_rate;
int input_frame_rate_starting_point_is_valid;
uint64_t frames_inward_measurement_start_time;
uint32_t frames_inward_frames_received_at_measurement_start_time;
uint64_t frames_inward_measurement_time;
uint32_t frames_inward_frames_received_at_measurement_time;
// other stuff...
pthread_t *player_thread;
abuf_t audio_buffer[BUFFER_FRAMES];
unsigned int max_frames_per_packet, input_num_channels, input_bit_depth, input_rate;
int input_bytes_per_frame, output_bytes_per_frame, output_sample_ratio;
int max_frame_size_change;
int64_t previous_random_number;
alac_file *decoder_info;
uint64_t packet_count;
uint64_t packet_count_since_flush;
int connection_state_to_output;
uint64_t first_packet_time_to_play;
int64_t time_since_play_started; // nanoseconds
// stats
uint64_t missing_packets, late_packets, too_late_packets, resend_requests;
int decoder_in_use;
// debug variables
int32_t last_seqno_read;
// mutexes and condition variables
pthread_cond_t flowcontrol;
pthread_mutex_t ab_mutex, flush_mutex, volume_control_mutex;
int fix_volume;
double initial_airplay_volume;
int initial_airplay_volume_set;
uint32_t timestamp_epoch, last_timestamp,
maximum_timestamp_interval; // timestamp_epoch of zero means not initialised, could start at 2
// or 1.
int ab_buffering, ab_synced;
int64_t first_packet_timestamp;
int flush_requested;
int flush_output_flushed; // true if the output device has been flushed.
uint32_t flush_rtp_timestamp;
uint64_t time_of_last_audio_packet;
seq_t ab_read, ab_write;
#ifdef CONFIG_MBEDTLS
mbedtls_aes_context dctx;
#endif
#ifdef CONFIG_POLARSSL
aes_context dctx;
#endif
#ifdef CONFIG_OPENSSL
AES_KEY aes;
#endif
int amountStuffed;
int32_t framesProcessedInThisEpoch;
int32_t framesGeneratedInThisEpoch;
int32_t correctionsRequestedInThisEpoch;
int64_t syncErrorsInThisEpoch;
// RTP stuff
// only one RTP session can be active at a time.
int rtp_running;
uint64_t rtp_time_of_last_resend_request_error_ns;
char client_ip_string[INET6_ADDRSTRLEN]; // the ip string of the client
uint16_t client_rtsp_port;
char self_ip_string[INET6_ADDRSTRLEN]; // the ip string being used by this program -- it
uint16_t self_rtsp_port; // could be one of many, so we need to know it
uint32_t self_scope_id; // if it's an ipv6 connection, this will be its scope
short connection_ip_family; // AF_INET / AF_INET6
SOCKADDR rtp_client_control_socket; // a socket pointing to the control port of the client
SOCKADDR rtp_client_timing_socket; // a socket pointing to the timing port of the client
int audio_socket; // our local [server] audio socket
int control_socket; // our local [server] control socket
int timing_socket; // local timing socket
uint16_t remote_control_port;
uint16_t remote_timing_port;
uint16_t local_audio_port;
uint16_t local_control_port;
uint16_t local_timing_port;
int64_t latency_delayed_timestamp; // this is for debugging only...
// this is what connects an rtp timestamp to the remote time
int udp_clock_is_initialised;
int udp_clock_sender_is_initialised;
int anchor_remote_info_is_valid;
// these can be modified if the master clock changes over time
uint64_t anchor_clock;
uint64_t anchor_time; // this is the time according to the clock
uint32_t anchor_rtptime;
// these are used to identify when the master clock becomes equal to the
// actual anchor clock information, so it can be used to avoid accumulating errors
uint64_t actual_anchor_clock;
uint64_t actual_anchor_time;
uint32_t actual_anchor_rtptime;
clock_status_t clock_status;
airplay_stream_c
airplay_stream_category; // is it a remote control stream or a normal "full service" stream?
// (will be unspecified if not build for AirPlay 2)
#ifdef CONFIG_AIRPLAY_2
char *airplay_gid; // UUID in the Bonjour advertisement -- if NULL, the group UUID is the same as
// the pi UUID
airplay_t airplay_type; // are we using AirPlay 1 or AirPlay 2 protocol on this connection?
airplay_stream_t airplay_stream_type; // is it realtime audio or buffered audio...
timing_t timing_type; // are we using NTP or PTP on this connection?
pthread_t *rtp_event_thread;
pthread_t *rtp_data_thread;
pthread_t rtp_ap2_control_thread;
pthread_t rtp_realtime_audio_thread;
pthread_t rtp_buffered_audio_thread;
int last_anchor_info_is_valid;
uint32_t last_anchor_rtptime;
uint64_t last_anchor_local_time;
uint64_t last_anchor_time_of_update;
uint64_t last_anchor_validity_start_time;
ssize_t ap2_audio_buffer_size;
ssize_t ap2_audio_buffer_minimum_size;
flush_request_t *flush_requests; // if non-null, there are flush requests, mutex protected
int ap2_flush_requested;
int ap2_flush_from_valid;
uint32_t ap2_flush_from_rtp_timestamp;
uint32_t ap2_flush_from_sequence_number;
uint32_t ap2_flush_until_rtp_timestamp;
uint32_t ap2_flush_until_sequence_number;
int ap2_rate; // protect with flush mutex, 0 means don't play, 1 means play
int ap2_play_enabled; // protect with flush mutex
ap2_pairing ap2_pairing_context;
int event_socket;
int data_socket;
SOCKADDR ap2_remote_control_socket_addr; // a socket pointing to the control port of the client
socklen_t ap2_remote_control_socket_addr_length;
int ap2_control_socket;
int realtime_audio_socket;
int buffered_audio_socket;
uint16_t local_data_port;
uint16_t local_event_port;
uint16_t local_ap2_control_port;
uint16_t local_realtime_audio_port;
uint16_t local_buffered_audio_port;
uint64_t audio_format;
uint64_t compression;
unsigned char *session_key; // needs to be free'd at the end
uint64_t frames_packet;
uint64_t type;
uint64_t networkTimeTimelineID; // the clock ID used by the player
uint8_t groupContainsGroupLeader; // information coming from the SETUP
#endif
// used as the initials values for calculating the rate at which the source thinks it's sending
// frames
uint32_t initial_reference_timestamp;
uint64_t initial_reference_time;
double remote_frame_rate;
// the ratio of the following should give us the operating rate, nominally 44,100
int64_t reference_to_previous_frame_difference;
uint64_t reference_to_previous_time_difference;
// debug variables
int request_sent;
int time_ping_count;
struct time_ping_record time_pings[time_ping_history];
uint64_t departure_time; // dangerous -- this assumes that there will never be two timing
// request in flight at the same time
pthread_mutex_t reference_time_mutex;
pthread_mutex_t watchdog_mutex;
double local_to_remote_time_gradient; // if no drift, this would be exactly 1.0; likely it's
// slightly above or below.
int local_to_remote_time_gradient_sample_count; // the number of samples used to calculate the
// gradient
// add the following to the local time to get the remote time modulo 2^64
uint64_t local_to_remote_time_difference; // used to switch between local and remote clocks
uint64_t local_to_remote_time_difference_measurement_time; // when the above was calculated
int last_stuff_request;
// int64_t play_segment_reference_frame;
// uint64_t play_segment_reference_frame_remote_time;
int32_t buffer_occupancy; // allow it to be negative because seq_diff may be negative
int64_t session_corrections;
int play_number_after_flush;
// remote control stuff. The port to which to send commands is not specified, so you have to use
// mdns to find it.
// at present, only avahi can do this
char *dacp_id; // id of the client -- used to find the port to be used
// uint16_t dacp_port; // port on the client to send remote control messages to, else
// zero
char *dacp_active_remote; // key to send to the remote controller
void *dapo_private_storage; // this is used for compatibility, if dacp stuff isn't enabled.
int enable_dither; // needed for filling silences before play actually starts
uint64_t dac_buffer_queue_minimum_length;
} rtsp_conn_info;
extern pthread_mutex_t playing_conn_lock;
extern int statistics_row; // will be reset to zero when debug level changes or statistics enabled
void reset_buffer(rtsp_conn_info *conn);
void get_audio_buffer_size_and_occupancy(unsigned int *size, unsigned int *occupancy,
rtsp_conn_info *conn);
int32_t modulo_32_offset(uint32_t from, uint32_t to);
void ab_resync(rtsp_conn_info *conn);
int player_prepare_to_play(rtsp_conn_info *conn);
int player_play(rtsp_conn_info *conn);
int player_stop(rtsp_conn_info *conn);
void player_volume(double f, rtsp_conn_info *conn);
void player_volume_without_notification(double f, rtsp_conn_info *conn);
void player_flush(uint32_t timestamp, rtsp_conn_info *conn);
// void player_full_flush(rtsp_conn_info *conn);
void player_put_packet(int original_format, seq_t seqno, uint32_t actual_timestamp, uint8_t *data,
int len, rtsp_conn_info *conn);
int64_t monotonic_timestamp(uint32_t timestamp,
rtsp_conn_info *conn); // add an epoch to the timestamp. The monotonic
// timestamp guaranteed to start between 2^32 2^33
// frames and continue up to 2^64 frames
// which is about 2*10^8 * 1,000 seconds at 384,000 frames per second -- about 2 trillion seconds.
// assumes, without checking, that successive timestamps in a series always span an interval of less
// than one minute.
#endif //_PLAYER_H