void player_put_packet(seq_t seqno, uint32_t actual_timestamp, int64_t timestamp, uint8_t *data,
int len, rtsp_conn_info *conn) {
- pthread_rwlock_rdlock(&conn->player_thread_lock);
- if (conn->player_thread != NULL) {
+ if (pthread_rwlock_tryrdlock(&conn->player_thread_lock) == 0) {
+ if (conn->player_thread != NULL) {
- // all timestamps are done at the output rate
- // the "actual_timestamp" is the one that comes in the packet, and is carried over for debugging
- // and checking only.
+ // all timestamps are done at the output rate
+ // the "actual_timestamp" is the one that comes in the packet, and is carried over for
+ // debugging
+ // and checking only.
- int64_t ltimestamp = timestamp * conn->output_sample_ratio;
+ int64_t ltimestamp = timestamp * conn->output_sample_ratio;
- // ignore a request to flush that has been made before the first packet...
- if (conn->packet_count == 0) {
- pthread_mutex_lock(&conn->flush_mutex);
- conn->flush_requested = 0;
- conn->flush_rtp_timestamp = 0;
- pthread_mutex_unlock(&conn->flush_mutex);
- }
+ // ignore a request to flush that has been made before the first packet...
+ if (conn->packet_count == 0) {
+ pthread_mutex_lock(&conn->flush_mutex);
+ conn->flush_requested = 0;
+ conn->flush_rtp_timestamp = 0;
+ pthread_mutex_unlock(&conn->flush_mutex);
+ }
- pthread_mutex_lock(&conn->ab_mutex);
- conn->packet_count++;
- conn->time_of_last_audio_packet = get_absolute_time_in_fp();
- if (conn->connection_state_to_output) { // if we are supposed to be processing these packets
+ pthread_mutex_lock(&conn->ab_mutex);
+ conn->packet_count++;
+ conn->time_of_last_audio_packet = get_absolute_time_in_fp();
+ if (conn->connection_state_to_output) { // if we are supposed to be processing these packets
- // if (flush_rtp_timestamp != 0)
- // debug(1,"Flush_rtp_timestamp is %u",flush_rtp_timestamp);
+ // if (flush_rtp_timestamp != 0)
+ // debug(1,"Flush_rtp_timestamp is %u",flush_rtp_timestamp);
- if ((conn->flush_rtp_timestamp != 0) && (ltimestamp <= conn->flush_rtp_timestamp)) {
- debug(3,
+ if ((conn->flush_rtp_timestamp != 0) && (ltimestamp <= conn->flush_rtp_timestamp)) {
+ debug(
+ 3,
"Dropping flushed packet in player_put_packet, seqno %u, timestamp %lld, flushing to "
"timestamp: %lld.",
seqno, ltimestamp, conn->flush_rtp_timestamp);
- } else {
- if ((conn->flush_rtp_timestamp != 0x0) &&
- (ltimestamp >
- conn->flush_rtp_timestamp)) // if we have gone past the flush boundary time
- conn->flush_rtp_timestamp = 0x0;
-
- abuf_t *abuf = 0;
-
- if (!conn->ab_synced) {
- debug(3, "syncing to seqno %u.", seqno);
- conn->ab_write = seqno;
- conn->ab_read = seqno;
- conn->ab_synced = 1;
- }
+ } else {
+ if ((conn->flush_rtp_timestamp != 0x0) &&
+ (ltimestamp >
+ conn->flush_rtp_timestamp)) // if we have gone past the flush boundary time
+ conn->flush_rtp_timestamp = 0x0;
+
+ abuf_t *abuf = 0;
+
+ if (!conn->ab_synced) {
+ debug(3, "syncing to seqno %u.", seqno);
+ conn->ab_write = seqno;
+ conn->ab_read = seqno;
+ conn->ab_synced = 1;
+ }
- // here, we should check for missing frames
- int resend_interval = (((250 * 44100) / 352) / 1000); // approximately 250 ms intervals
- const int number_of_resend_attempts = 8;
- int latency_based_resend_interval =
- (conn->latency) / (number_of_resend_attempts * conn->max_frames_per_packet);
- if (latency_based_resend_interval > resend_interval)
- resend_interval = latency_based_resend_interval;
-
- if (conn->resend_interval != resend_interval) {
- debug(2, "Resend interval for latency of %" PRId64 " frames is %d frames.", conn->latency,
- resend_interval);
- conn->resend_interval = resend_interval;
- }
- if (!conn->ab_buffering) {
- int j;
- for (j = 1; j <= number_of_resend_attempts; j++) {
- // check j times, after a short period of has elapsed, assuming 352 frames per packet
-
- int back_step = resend_interval * j;
-
- int32_t sd = seq_diff(conn->ab_read, conn->ab_write, conn->ab_read);
- int k;
- for (k = -2; k <= 2; k++) {
- if ((back_step + k) < sd) { // if it's within the range of frames in use...
- int item_to_check = (conn->ab_write - (back_step + k)) & 0xffff;
- seq_t next = item_to_check;
- abuf_t *check_buf = conn->audio_buffer + BUFIDX(next);
- if ((!check_buf->ready) &&
- (check_buf->resend_level <
- j)) { // prevent multiple requests from the same level of lookback
- check_buf->resend_level = j;
- if (config.disable_resend_requests == 0) {
- rtp_request_resend(next, 1, conn);
- if ((back_step + k + resend_interval) >= sd)
- debug(2, "Last-ditch (#%d) resend request for packet %u in range %u to %u. "
- "Looking back %d packets.",
- j, next, conn->ab_read, conn->ab_write, back_step + k);
- conn->resend_requests++;
+ // here, we should check for missing frames
+ int resend_interval = (((250 * 44100) / 352) / 1000); // approximately 250 ms intervals
+ const int number_of_resend_attempts = 8;
+ int latency_based_resend_interval =
+ (conn->latency) / (number_of_resend_attempts * conn->max_frames_per_packet);
+ if (latency_based_resend_interval > resend_interval)
+ resend_interval = latency_based_resend_interval;
+
+ if (conn->resend_interval != resend_interval) {
+ debug(2, "Resend interval for latency of %" PRId64 " frames is %d frames.",
+ conn->latency, resend_interval);
+ conn->resend_interval = resend_interval;
+ }
+ if (!conn->ab_buffering) {
+ int j;
+ for (j = 1; j <= number_of_resend_attempts; j++) {
+ // check j times, after a short period of has elapsed, assuming 352 frames per packet
+
+ int back_step = resend_interval * j;
+
+ int32_t sd = seq_diff(conn->ab_read, conn->ab_write, conn->ab_read);
+ int k;
+ for (k = -2; k <= 2; k++) {
+ if ((back_step + k) < sd) { // if it's within the range of frames in use...
+ int item_to_check = (conn->ab_write - (back_step + k)) & 0xffff;
+ seq_t next = item_to_check;
+ abuf_t *check_buf = conn->audio_buffer + BUFIDX(next);
+ if ((!check_buf->ready) &&
+ (check_buf->resend_level <
+ j)) { // prevent multiple requests from the same level of lookback
+ check_buf->resend_level = j;
+ if (config.disable_resend_requests == 0) {
+ rtp_request_resend(next, 1, conn);
+ if ((back_step + k + resend_interval) >= sd)
+ debug(2, "Last-ditch (#%d) resend request for packet %u in range %u to %u. "
+ "Looking back %d packets.",
+ j, next, conn->ab_read, conn->ab_write, back_step + k);
+ conn->resend_requests++;
+ }
}
}
}
}
}
- }
- if (conn->ab_write == seqno) { // expected packet
- abuf = conn->audio_buffer + BUFIDX(seqno);
- conn->ab_write = SUCCESSOR(seqno);
- } else if (seq_order(conn->ab_write, seqno, conn->ab_read)) { // newer than expected
- // if (ORDINATE(seqno)>(BUFFER_FRAMES*7)/8)
- // debug(1,"An interval of %u frames has opened, with ab_read: %u, ab_write: %u and seqno:
- // %u.",seq_diff(ab_read,seqno),ab_read,ab_write,seqno);
- int32_t gap = seq_diff(conn->ab_write, seqno, conn->ab_read);
- if (gap <= 0)
- debug(1, "Unexpected gap size: %d.", gap);
- int i;
- for (i = 0; i < gap; i++) {
- abuf = conn->audio_buffer + BUFIDX(seq_sum(conn->ab_write, i));
- abuf->ready = 0; // to be sure, to be sure
- abuf->resend_level = 0;
- abuf->timestamp = 0;
- abuf->given_timestamp = 0;
- abuf->sequence_number = 0;
+ if (conn->ab_write == seqno) { // expected packet
+ abuf = conn->audio_buffer + BUFIDX(seqno);
+ conn->ab_write = SUCCESSOR(seqno);
+ } else if (seq_order(conn->ab_write, seqno, conn->ab_read)) { // newer than expected
+ // if (ORDINATE(seqno)>(BUFFER_FRAMES*7)/8)
+ // debug(1,"An interval of %u frames has opened, with ab_read: %u, ab_write: %u and
+ // seqno:
+ // %u.",seq_diff(ab_read,seqno),ab_read,ab_write,seqno);
+ int32_t gap = seq_diff(conn->ab_write, seqno, conn->ab_read);
+ if (gap <= 0)
+ debug(1, "Unexpected gap size: %d.", gap);
+ int i;
+ for (i = 0; i < gap; i++) {
+ abuf = conn->audio_buffer + BUFIDX(seq_sum(conn->ab_write, i));
+ abuf->ready = 0; // to be sure, to be sure
+ abuf->resend_level = 0;
+ abuf->timestamp = 0;
+ abuf->given_timestamp = 0;
+ abuf->sequence_number = 0;
+ }
+ // debug(1,"N %d s %u.",seq_diff(ab_write,PREDECESSOR(seqno))+1,ab_write);
+ abuf = conn->audio_buffer + BUFIDX(seqno);
+ // rtp_request_resend(ab_write, gap);
+ // resend_requests++;
+ conn->ab_write = SUCCESSOR(seqno);
+ } else if (seq_order(conn->ab_read, seqno, conn->ab_read)) { // late but not yet played
+ conn->late_packets++;
+ abuf = conn->audio_buffer + BUFIDX(seqno);
+ /*
+ if (abuf->ready)
+ debug(1,"Late apparently duplicate packet received that is %d packets
+ late.",seq_diff(seqno, conn->ab_write, conn->ab_read));
+ else
+ debug(1,"Late packet received that is %d packets late.",seq_diff(seqno,
+ conn->ab_write, conn->ab_read));
+ */
+ } else { // too late.
+
+ // debug(1,"Too late packet received that is %d packets late.",seq_diff(seqno,
+ // conn->ab_write, conn->ab_read));
+ conn->too_late_packets++;
}
- // debug(1,"N %d s %u.",seq_diff(ab_write,PREDECESSOR(seqno))+1,ab_write);
- abuf = conn->audio_buffer + BUFIDX(seqno);
- // rtp_request_resend(ab_write, gap);
- // resend_requests++;
- conn->ab_write = SUCCESSOR(seqno);
- } else if (seq_order(conn->ab_read, seqno, conn->ab_read)) { // late but not yet played
- conn->late_packets++;
- abuf = conn->audio_buffer + BUFIDX(seqno);
- /*
- if (abuf->ready)
- debug(1,"Late apparently duplicate packet received that is %d packets
- late.",seq_diff(seqno, conn->ab_write, conn->ab_read));
- else
- debug(1,"Late packet received that is %d packets late.",seq_diff(seqno,
- conn->ab_write, conn->ab_read));
- */
- } else { // too late.
-
- // debug(1,"Too late packet received that is %d packets late.",seq_diff(seqno,
- // conn->ab_write, conn->ab_read));
- conn->too_late_packets++;
- }
- // pthread_mutex_unlock(&ab_mutex);
-
- if (abuf) {
- int datalen = conn->max_frames_per_packet;
- if (alac_decode(abuf->data, &datalen, data, len, conn) == 0) {
- abuf->ready = 1;
- abuf->length = datalen;
- abuf->timestamp = ltimestamp;
- abuf->given_timestamp = actual_timestamp;
- abuf->sequence_number = seqno;
- } else {
- debug(1, "Bad audio packet detected and discarded.");
- abuf->ready = 0;
- abuf->resend_level = 0;
- abuf->timestamp = 0;
- abuf->given_timestamp = 0;
- abuf->sequence_number = 0;
+ // pthread_mutex_unlock(&ab_mutex);
+
+ if (abuf) {
+ int datalen = conn->max_frames_per_packet;
+ if (alac_decode(abuf->data, &datalen, data, len, conn) == 0) {
+ abuf->ready = 1;
+ abuf->length = datalen;
+ abuf->timestamp = ltimestamp;
+ abuf->given_timestamp = actual_timestamp;
+ abuf->sequence_number = seqno;
+ } else {
+ debug(1, "Bad audio packet detected and discarded.");
+ abuf->ready = 0;
+ abuf->resend_level = 0;
+ abuf->timestamp = 0;
+ abuf->given_timestamp = 0;
+ abuf->sequence_number = 0;
+ }
}
- }
- // pthread_mutex_lock(&ab_mutex);
+ // pthread_mutex_lock(&ab_mutex);
+ }
+ int rc = pthread_cond_signal(&conn->flowcontrol);
+ if (rc)
+ debug(1, "Error signalling flowcontrol.");
}
- int rc = pthread_cond_signal(&conn->flowcontrol);
- if (rc)
- debug(1, "Error signalling flowcontrol.");
+ pthread_mutex_unlock(&conn->ab_mutex);
+ } else {
+ debug(
+ 1,
+ "No player thread while adding a player_put_packet calle was made -- packet discarded.");
}
- pthread_mutex_unlock(&conn->ab_mutex);
+ pthread_rwlock_unlock(&conn->player_thread_lock);
} else {
- debug(1,
- "No player thread while adding a player_put_packet calle was made -- packet discarded.");
+ debug(1, "player_put_packet discarded packet %d because the player thread was locked.", seqno);
}
- pthread_rwlock_unlock(&conn->player_thread_lock);
}
int32_t rand_in_range(int32_t exclusive_range_limit) {
}
void rtp_audio_receiver_cleanup_handler(void *arg) {
+ debug(2, "Audio Receiver Cleanup.");
rtsp_conn_info *conn = (rtsp_conn_info *)arg;
close(conn->audio_socket);
- debug(3,"Audio Receiver Cleanup Successful.");
+ debug(2, "Audio Receiver Cleanup Successful.");
}
void *rtp_audio_receiver(void *arg) {
ssize_t nread;
while (1) {
-
nread = recv(conn->audio_socket, packet, sizeof(packet), 0);
uint64_t local_time_now_fp = get_absolute_time_in_fp();
}
void rtp_control_handler_cleanup_handler(void *arg) {
+ debug(2, "Control Receiver Cleanup.");
rtsp_conn_info *conn = (rtsp_conn_info *)arg;
close(conn->control_socket);
- debug(3,"Control Receiver Cleanup Successful.");
+ debug(2, "Control Receiver Cleanup Successful.");
}
void *rtp_control_receiver(void *arg) {
pthread_exit(NULL);
}
-pthread_t timer_requester;
-
void rtp_timing_receiver_cleanup_handler(void *arg) {
+ debug(2, "Timing Receiver Cleanup.");
rtsp_conn_info *conn = (rtsp_conn_info *)arg;
- pthread_cancel(timer_requester);
- pthread_join(timer_requester, NULL);
+ pthread_cancel(conn->timer_requester);
+ pthread_join(conn->timer_requester, NULL);
close(conn->timing_socket);
- debug(3,"Timing Receiver Cleanup Successful.");
+ debug(2, "Timing Receiver Cleanup Successful.");
}
void *rtp_timing_receiver(void *arg) {
uint8_t packet[2048];
ssize_t nread;
- pthread_create(&timer_requester, NULL, &rtp_timing_sender, arg);
+ pthread_create(&conn->timer_requester, NULL, &rtp_timing_sender, arg);
// struct timespec att;
uint64_t distant_receive_time, distant_transmit_time, arrival_time, return_time;
local_to_remote_time_jitters = 0;
return_time = arrival_time - conn->departure_time;
- //uint64_t rtus = (return_time * 1000000) >> 32;
+ // uint64_t rtus = (return_time * 1000000) >> 32;
if (((return_time * 1000000) >> 32) < 300000) {
struct sockaddr_in *sa = (struct sockaddr_in *)&local;
sport = ntohs(sa->sin_port);
}
- // fcntl(local_socket, F_SETFL, O_NONBLOCK);
-
- // don't wait if outputting something will take longer than this time.
- struct timeval timeout;
- timeout.tv_sec = 0;
- timeout.tv_usec = 2000; // two milliseconds
-
- if (setsockopt(local_socket, SOL_SOCKET, SO_SNDTIMEO, (char *)&timeout, sizeof(timeout)) < 0)
- debug(1, "bind_port set output timeout failed.");
- // else
- // debug(1,"bind_port set output timeout succeeded.");
-
*sock = local_socket;
return sport;
}
projects / thirdparty / shairport-sync.git / commitdiff
