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Move to using pthread_cancel in place of pthread_kill -- due to a problem in CYGWIN, but the move makes the code simpler and more rugged.

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This commit is contained in:
Mike Brady
2018-05-17 10:16:44 +01:00
parent 9aa920ab81
commit 7d0eb37958
5 changed files with 100 additions and 160 deletions
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mdns.c
+2 -2
View File
@@ -113,7 +113,7 @@ void *mdns_dacp_monitor(char *dacp_id) {
debug(1, "Error starting a DACP monitor.");
}
} else
debug(1, "Can't start a DACP monitor -- none registered.");
debug(3, "Can't start a DACP monitor -- none registered.");
}
return reply;
}
@@ -122,7 +122,7 @@ void mdns_dacp_dont_monitor(void *userdata) {
if ((config.mdns) && (config.mdns->mdns_dacp_dont_monitor)) {
config.mdns->mdns_dacp_dont_monitor(userdata);
} else
debug(1, "Can't stop a DACP monitor -- none registered.");
debug(3, "Can't stop a DACP monitor -- none registered.");
}
void mdns_ls_backends(void) {
mdns_backend **b = NULL;
player.c
+8 -16
View File
@@ -1449,7 +1449,6 @@ static void *player_thread_func(void *arg) {
// it's safe now
conn->please_stop = 0;
conn->packet_count = 0;
conn->previous_random_number = 0;
conn->input_bytes_per_frame = 4;
@@ -2305,7 +2304,7 @@ static void *player_thread_func(void *arg) {
}
}
debug(3, "Play thread main loop exited.");
debug(3, "Connection %d: player thread main loop exit.", conn->connection_number);
if (config.statistics_requested) {
int rawSeconds = (int)difftime(time(NULL), playstart);
@@ -2327,23 +2326,16 @@ static void *player_thread_func(void *arg) {
if (config.output->stop)
config.output->stop();
debug(2, "Shutting down audio, control and timing threads");
conn->please_stop = 1;
pthread_kill(rtp_audio_thread, SIGUSR1);
pthread_kill(rtp_control_thread, SIGUSR1);
pthread_kill(rtp_timing_thread, SIGUSR1);
debug(3, "Shutting down timing, control and audio threads");
pthread_cancel(rtp_timing_thread);
pthread_join(rtp_timing_thread, NULL);
debug(2, "timing thread joined");
pthread_join(rtp_audio_thread, NULL);
debug(2, "audio thread joined");
pthread_cancel(rtp_control_thread);
pthread_join(rtp_control_thread, NULL);
debug(2, "control thread joined");
pthread_cancel(rtp_audio_thread);
pthread_join(rtp_audio_thread, NULL);
clear_reference_timestamp(conn);
conn->rtp_running = 0;
// usleep(100000); // allow this time to (?) allow the alsa subsystem to finish cleaning up after
// itself. 50 ms seems too short
debug(3, "Freeing audio buffers and decoders.");
free_audio_buffers(conn);
@@ -2362,7 +2354,7 @@ static void *player_thread_func(void *arg) {
if (rc)
debug(1, "Error destroying vol_mutex variable.");
debug(2, "Player thread exit on RTSP conversation thread %d.", conn->connection_number);
debug(2, "Connection %d: player thread terminated.", conn->connection_number);
if (conn->dacp_id) {
free(conn->dacp_id);
conn->dacp_id = NULL;
@@ -2373,7 +2365,7 @@ static void *player_thread_func(void *arg) {
free(tbuf);
if (sbuf)
free(sbuf);
return 0;
pthread_exit(NULL);
}
// takes the volume as specified by the airplay protocol
player.h
-2
View File
@@ -93,7 +93,6 @@ typedef struct {
int max_frame_size_change;
int64_t previous_random_number;
alac_file *decoder_info;
uint32_t please_stop;
uint64_t packet_count;
int connection_state_to_output;
int player_thread_please_stop;
@@ -180,7 +179,6 @@ typedef struct {
uint64_t local_to_remote_time_difference; // used to switch between local and remote clocks
int timing_sender_stop; // for asking the timing-sending thread to stop
int last_stuff_request;
int64_t play_segment_reference_frame;
rtp.c
+88 -137
View File
@@ -48,8 +48,6 @@
uint64_t local_to_remote_time_jitters;
uint64_t local_to_remote_time_jitters_count;
void memory_barrier();
void rtp_initialise(rtsp_conn_info *conn) {
conn->rtp_time_of_last_resend_request_error_fp = 0;
conn->rtp_running = 0;
@@ -67,10 +65,13 @@ void rtp_terminate(rtsp_conn_info *conn) {
debug(1, "Error destroying reference_time_mutex variable.");
}
void *rtp_audio_receiver(void *arg) {
debug(3, "Audio receiver -- Server RTP thread starting.");
void rtp_audio_receiver_cleanup_handler(void *arg) {
rtsp_conn_info *conn = (rtsp_conn_info *)arg;
close(conn->audio_socket);
}
// we inherit the signal mask (SIGUSR1)
void *rtp_audio_receiver(void *arg) {
pthread_cleanup_push(rtp_audio_receiver_cleanup_handler, arg);
rtsp_conn_info *conn = (rtsp_conn_info *)arg;
int32_t last_seqno = -1;
@@ -87,17 +88,8 @@ void *rtp_audio_receiver(void *arg) {
float stat_M2 = 0.0;
ssize_t nread;
while (conn->please_stop == 0) {
fd_set readfds;
FD_ZERO(&readfds);
FD_SET(conn->audio_socket, &readfds);
do {
memory_barrier();
} while (conn->please_stop == 0 &&
pselect(conn->audio_socket + 1, &readfds, NULL, NULL, NULL, &pselect_sigset) <= 0);
if (conn->please_stop != 0) {
break;
}
while (1) {
nread = recv(conn->audio_socket, packet, sizeof(packet), 0);
uint64_t local_time_now_fp = get_absolute_time_in_fp();
@@ -181,17 +173,24 @@ void *rtp_audio_receiver(void *arg) {
}
}
/*
debug(3, "Audio receiver -- Server RTP thread interrupted. terminating.");
close(conn->audio_socket);
*/
debug(1, "Audio receiver thread \"normal\" exit -- this can't happen. Hah!");
pthread_cleanup_pop(0); // don't execute anything here.
debug(2, "Audio receiver thread exit.");
pthread_exit(NULL);
}
return NULL;
void rtp_control_handler_cleanup_handler(void *arg) {
rtsp_conn_info *conn = (rtsp_conn_info *)arg;
close(conn->control_socket);
}
void *rtp_control_receiver(void *arg) {
// we inherit the signal mask (SIGUSR1)
debug(3, "Control receiver -- Server RTP thread starting.");
pthread_cleanup_push(rtp_control_handler_cleanup_handler, arg);
rtsp_conn_info *conn = (rtsp_conn_info *)arg;
conn->reference_timestamp = 0; // nothing valid received yet
@@ -200,18 +199,7 @@ void *rtp_control_receiver(void *arg) {
uint64_t remote_time_of_sync;
int64_t sync_rtp_timestamp;
ssize_t nread;
while (conn->please_stop == 0) {
fd_set readfds;
FD_ZERO(&readfds);
FD_SET(conn->control_socket, &readfds);
do {
memory_barrier();
} while (conn->please_stop == 0 &&
pselect(conn->control_socket + 1, &readfds, NULL, NULL, NULL, &pselect_sigset) <= 0);
if (conn->please_stop != 0) {
break;
}
while (1) {
nread = recv(conn->control_socket, packet, sizeof(packet), 0);
// local_time_now = get_absolute_time_in_fp();
// clock_gettime(CLOCK_MONOTONIC,&tn);
@@ -223,50 +211,50 @@ void *rtp_control_receiver(void *arg) {
(drand48() > config.diagnostic_drop_packet_fraction)) {
ssize_t plen = nread;
if (packet[1] == 0xd4) { // sync data
/*
// the following stanza is for debugging only -- normally commented out.
{
char obf[4096];
char *obfp = obf;
int obfc;
for (obfc = 0; obfc < plen; obfc++) {
snprintf(obfp, 3, "%02X", packet[obfc]);
obfp += 2;
};
*obfp = 0;
// get raw timestamp information
// I think that a good way to understand these timestamps is that
// (1) the rtlt below is the timestamp of the frame that should be playing at the
client-time specified in the packet if there was no delay
// and (2) that the rt below is the timestamp of the frame that should be playing
at
the client-time specified in the packet on this device taking account of the delay
// Thus, (3) the latency can be calculated by subtracting the second from the
first.
// There must be more to it -- there something missing.
// In addition, it seems that if the value of the short represented by the second
pair
of bytes in the packe is 7
// then an extra time lag is expected to be added, presumably by the AirPort
Express.
Best guess is that this delay is 11,025 frames.
// uint32_t rtlt = nctohl(&packet[4]); // raw timestamp less latency
// uint32_t rt = nctohl(&packet[16]); // raw timestamp
// uint32_t fl = nctohs(&packet[2]); //
// debug(1,"Sync Packet of %d bytes received: \"%s\", flags: %d, timestamps %u and
%u,
giving a latency of %d frames.",plen,obf,fl,rt,rtlt,rt-rtlt);
// debug(1,"Monotonic timestamps are: %" PRId64 " and %" PRId64 "
respectively.",monotonic_timestamp(rt, conn),monotonic_timestamp(rtlt, conn));
}
*/
if (packet[1] == 0xd4) { // sync data
/*
// the following stanza is for debugging only -- normally commented out.
{
char obf[4096];
char *obfp = obf;
int obfc;
for (obfc = 0; obfc < plen; obfc++) {
snprintf(obfp, 3, "%02X", packet[obfc]);
obfp += 2;
};
*obfp = 0;
// get raw timestamp information
// I think that a good way to understand these timestamps is that
// (1) the rtlt below is the timestamp of the frame that should be playing at the
// client-time specified in the packet if there was no delay
// and (2) that the rt below is the timestamp of the frame that should be playing
// at the client-time specified in the packet on this device taking account of
// the delay
// Thus, (3) the latency can be calculated by subtracting the second from the
// first.
// There must be more to it -- there something missing.
// In addition, it seems that if the value of the short represented by the second
// pair of bytes in the packe is 7
// then an extra time lag is expected to be added, presumably by
// the AirPort Express.
// Best guess is that this delay is 11,025 frames.
// uint32_t rtlt = nctohl(&packet[4]); // raw timestamp less latency
// uint32_t rt = nctohl(&packet[16]); // raw timestamp
// uint32_t fl = nctohs(&packet[2]); //
// debug(1,"Sync Packet of %d bytes received: \"%s\", flags: %d, timestamps %u and
%u,
giving a latency of %d frames.",plen,obf,fl,rt,rtlt,rt-rtlt);
// debug(1,"Monotonic timestamps are: %" PRId64 " and %" PRId64 "
respectively.",monotonic_timestamp(rt, conn),monotonic_timestamp(rtlt, conn));
}
*/
if (conn->local_to_remote_time_difference) { // need a time packet to be interchanged
// first...
@@ -406,15 +394,13 @@ void *rtp_control_receiver(void *arg) {
debug(1, "Control Receiver -- error receiving a packet.");
}
}
debug(3, "Control RTP thread interrupted. terminating.");
close(conn->control_socket);
return NULL;
debug(1, "Control RTP thread \"normal\" exit -- this can't happen. Hah!");
pthread_cleanup_pop(0); // don't execute anything here.
debug(2, "Control RTP thread exit.");
pthread_exit(NULL);
}
void *rtp_timing_sender(void *arg) {
debug(3, "Timing sender thread starting.");
rtsp_conn_info *conn = (rtsp_conn_info *)arg;
struct timing_request {
char leader;
@@ -434,9 +420,7 @@ void *rtp_timing_sender(void *arg) {
req.seqno = htons(7);
conn->time_ping_count = 0;
// we inherit the signal mask (SIGUSR1)
while (conn->timing_sender_stop == 0) {
while (1) {
// debug(1,"Send a timing request");
if (!conn->rtp_running)
@@ -456,17 +440,6 @@ void *rtp_timing_sender(void *arg) {
msgsize = sizeof(struct sockaddr_in6);
}
#endif
fd_set writefds;
FD_ZERO(&writefds);
FD_SET(conn->timing_socket, &writefds);
do {
memory_barrier();
} while (conn->timing_sender_stop == 0 &&
pselect(conn->timing_socket + 1, NULL, &writefds, NULL, NULL, &pselect_sigset) <= 0);
if (conn->timing_sender_stop != 0) {
break;
}
if ((config.diagnostic_drop_packet_fraction == 0.0) ||
(drand48() > config.diagnostic_drop_packet_fraction)) {
if (sendto(conn->timing_socket, &req, sizeof(req), 0,
@@ -481,35 +454,30 @@ void *rtp_timing_sender(void *arg) {
request_number++;
// this is to deal with the possibility of missing a timing_sender_stop signal.
// if the signal came in just before the usleep, then it wouldn't cause the sleep to end.
// so, we will wait a maximum time of the wait_interval
int wait_time;
int wait_interval = 20000; // 20 milliseconds
if (request_number <= 4)
wait_time = 500000;
usleep(500000); //these are thread cancellation points
else
wait_time = 3000000;
while ((wait_time > 0) && (conn->timing_sender_stop == 0)) {
usleep(wait_interval);
wait_time -= wait_interval;
}
usleep(3000000);
}
debug(3, "rtp_timing_sender thread interrupted. terminating.");
return NULL;
pthread_exit(NULL);
}
pthread_t timer_requester;
void rtp_timing_receiver_cleanup_handler(void *arg) {
rtsp_conn_info *conn = (rtsp_conn_info *)arg;
pthread_cancel(timer_requester);
pthread_join(timer_requester, NULL);
close(conn->timing_socket);
}
void *rtp_timing_receiver(void *arg) {
debug(3, "Timing receiver -- Server RTP thread starting.");
// we inherit the signal mask (SIGUSR1)
pthread_cleanup_push(rtp_timing_receiver_cleanup_handler, arg);
rtsp_conn_info *conn = (rtsp_conn_info *)arg;
uint8_t packet[2048];
ssize_t nread;
conn->timing_sender_stop = 0;
pthread_t timer_requester;
pthread_create(&timer_requester, NULL, &rtp_timing_sender, arg);
// struct timespec att;
uint64_t distant_receive_time, distant_transmit_time, arrival_time, return_time;
@@ -521,17 +489,7 @@ void *rtp_timing_receiver(void *arg) {
uint64_t first_local_to_remote_time_difference = 0;
// uint64_t first_local_to_remote_time_difference_time;
// uint64_t l2rtd = 0;
while (conn->please_stop == 0) {
fd_set readfds;
FD_ZERO(&readfds);
FD_SET(conn->timing_socket, &readfds);
do {
memory_barrier();
} while (conn->please_stop == 0 &&
pselect(conn->timing_socket + 1, &readfds, NULL, NULL, NULL, &pselect_sigset) <= 0);
if (conn->please_stop != 0) {
break;
}
while (1) {
nread = recv(conn->timing_socket, packet, sizeof(packet), 0);
if (nread >= 0) {
@@ -738,17 +696,10 @@ void *rtp_timing_receiver(void *arg) {
}
}
debug(3, "Timing thread interrupted. terminating.");
conn->timing_sender_stop = 1;
void *retval;
pthread_kill(timer_requester, SIGUSR1);
debug(3, "Wait for timer requester to exit.");
pthread_join(timer_requester, &retval);
debug(3, "Closed and terminated timer requester thread.");
debug(3, "Timing RTP thread terminated.");
close(conn->timing_socket);
return NULL;
debug(1, "Timing Receiver RTP thread \"normal\" exit -- this can't happen. Hah!");
pthread_cleanup_pop(0); // don't execute anything here.
debug(2, "Timing Receiver RTP thread exit.");
pthread_exit(NULL);
}
static uint16_t bind_port(int ip_family, const char *self_ip_address, uint32_t scope_id,
rtsp.c
+2 -3
View File
@@ -1972,8 +1972,7 @@ static void *rtsp_conversation_thread_func(void *pconn) {
playing_conn = NULL;
pthread_mutex_unlock(&play_lock);
}
debug(2, "RTSP conversation thread %d terminated.", conn->connection_number);
// please_shutdown = 0;
debug(2, "Connection %d: RTSP thread terminated.", conn->connection_number);
conn->running = 0;
// release the player_thread_lock
@@ -1982,7 +1981,7 @@ static void *rtsp_conversation_thread_func(void *pconn) {
debug(1, "Error %d destroying player_thread_lock for conversation thread %d.", rwld,
conn->connection_number);
return NULL;
pthread_exit(NULL);
}
/*