nils/shairport-sync
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Only request a resend one time, after about a second. Stop making resend requests if there is a send-to error for 10 seconds.

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This commit is contained in:
Mike Brady
2018-02-10 17:43:47 +00:00
parent 419fe95a56
commit f94d6a5d16
3 changed files with 36 additions and 17 deletions
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player.c
+10 -6
View File
@@ -142,7 +142,6 @@ int64_t monotonic_timestamp(uint32_t timestamp, rtsp_conn_info *conn) {
conn->timestamp_epoch--;
}
}
conn->last_timestamp = timestamp;
return_value = conn->timestamp_epoch;
return_value <<= 32;
return_value += timestamp;
@@ -155,6 +154,7 @@ int64_t monotonic_timestamp(uint32_t timestamp, rtsp_conn_info *conn) {
}
if (return_value < 0)
debug(1, "monotonic rtptime is negative!");
conn->last_timestamp = timestamp;
return return_value;
}
@@ -870,8 +870,8 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) {
if (local_time_now >= conn->first_packet_time_to_play) {
debug(
1,
"First packet is late! It should have played before now. Flushing 0.1 seconds");
player_flush(conn->first_packet_timestamp + 4410 * conn->output_sample_ratio, conn);
"First packet is late! It should have played before now. Flushing 0.5 seconds");
player_flush(conn->first_packet_timestamp + 5 * 4410 * conn->output_sample_ratio, conn);
}
}
}
@@ -1155,7 +1155,11 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) {
// packets have arrived... last-chance resend
if (!conn->ab_buffering) {
for (i = 64; i < (seq_diff(conn->ab_read, conn->ab_write, conn->ab_read) / 2); i = (i * 2)) {
// check at half second intervals
// for (i = 500000/7982 ; i < (seq_diff(conn->ab_read, conn->ab_write, conn->ab_read)); i += 500000/7982) {
// check once, after a second has elapsed, assuming 352 frames per packet
i = 44100/352;
if (i<seq_diff(conn->ab_read, conn->ab_write, conn->ab_read)) {
seq_t next = seq_sum(conn->ab_read, i);
abuf = conn->audio_buffer + BUFIDX(next);
if (!abuf->ready) {
@@ -1420,7 +1424,7 @@ static void *player_thread_func(void *arg) {
conn->timestamp_epoch = 0; // indicate that the next timestamp will be the first one.
conn->maximum_timestamp_interval =
conn->input_rate * 60; // actually there shouldn't be more than about 13v
conn->input_rate * 60; // actually there shouldn't be more than about 13
// seconds of a gap between successive rtptimes, at
// worst
@@ -1429,7 +1433,7 @@ static void *player_thread_func(void *arg) {
// debug(1, "Output sample ratio is %d.", conn->output_sample_ratio);
conn->max_frame_size_change =
500 * conn->output_sample_ratio; // we add or subtract one frame at the nominal
1 * conn->output_sample_ratio; // we add or subtract one frame at the nominal
// rate, multiply it by the frame ratio.
// but, on some occasions, more than one frame could be added
player.h
+2
View File
@@ -129,6 +129,8 @@ typedef struct {
// RTP stuff
// only one RTP session can be active at a time.
int rtp_running;
uint64_t rtp_time_of_last_resend_request_error_fp;
char client_ip_string[INET6_ADDRSTRLEN]; // the ip string pointing to the client
char self_ip_string[INET6_ADDRSTRLEN]; // the ip string being used by this program -- it
rtp.c
+24 -11
View File
@@ -46,7 +46,7 @@
void memory_barrier();
void rtp_initialise(rtsp_conn_info *conn) {
conn->rtp_time_of_last_resend_request_error_fp = 0;
conn->rtp_running = 0;
// initialise the timer mutex
int rc = pthread_mutex_init(&conn->reference_time_mutex, NULL);
@@ -131,7 +131,7 @@ void *rtp_audio_receiver(void *arg) {
pktp += 4;
plen -= 4;
}
seq_t seqno = ntohs(*(unsigned short *)(pktp + 2));
seq_t seqno = ntohs(*(uint16_t *)(pktp + 2));
// increment last_seqno and see if it's the same as the incoming seqno
if (last_seqno == -1)
@@ -142,7 +142,7 @@ void *rtp_audio_receiver(void *arg) {
// debug(3, "RTP: Packets out of sequence: expected: %d, got %d.", last_seqno, seqno);
last_seqno = seqno; // reset warning...
}
int64_t timestamp = monotonic_timestamp(ntohl(*(unsigned long *)(pktp + 4)), conn);
int64_t timestamp = monotonic_timestamp(ntohl(*(uint32_t*)(pktp + 4)), conn);
// if (packet[1]&0x10)
// debug(1,"Audio packet Extension bit set.");
@@ -224,6 +224,8 @@ void *rtp_control_receiver(void *arg) {
// debug(1,"Remote Sync Time: %0llx.",remote_time_of_sync);
sync_rtp_timestamp = monotonic_timestamp(ntohl(*((uint32_t *)&packet[16])), conn);
debug(1,"Sync timestamp is %u.",ntohl(*((uint32_t *)&packet[16])));
if (config.userSuppliedLatency) {
if (config.userSuppliedLatency != conn->latency) {
@@ -232,6 +234,7 @@ void *rtp_control_receiver(void *arg) {
conn->latency = config.userSuppliedLatency;
} else if (packet[0] &
0x10) { // only set latency if it's a packet just after a flush or resume
debug(1,"Set latency");
int64_t rtp_timestamp_less_latency =
monotonic_timestamp(ntohl(*((uint32_t *)&packet[4])), conn);
int64_t la = sync_rtp_timestamp - rtp_timestamp_less_latency + config.fixedLatencyOffset;
@@ -277,12 +280,12 @@ void *rtp_control_receiver(void *arg) {
debug(1, "Sync packet received before we got a timing packet back.");
}
} else if (packet[1] == 0xd6) { // resent audio data in the control path -- whaale only?
// debug(1, "Control Port -- Retransmitted Audio Data Packet received.");
debug(1, "Control Port -- Retransmitted Audio Data Packet received.");
pktp = packet + 4;
plen -= 4;
seq_t seqno = ntohs(*(unsigned short *)(pktp + 2));
seq_t seqno = ntohs(*(uint16_t*)(pktp + 2));
int64_t timestamp = monotonic_timestamp(ntohl(*(unsigned long *)(pktp + 4)), conn);
int64_t timestamp = monotonic_timestamp(ntohl(*(uint32_t*)(pktp + 4)), conn);
pktp += 12;
plen -= 12;
@@ -446,9 +449,12 @@ void *rtp_timing_receiver(void *arg) {
return_time = arrival_time - conn->departure_time;
uint64_t rtus = (return_time * 1000000) >> 32;
if (rtus < 300000) {
// debug(2,"Synchronisation ping return time is %f milliseconds.",(rtus*1.0)/1000);
// distant_receive_time =
// ((uint64_t)ntohl(*((uint32_t*)&packet[16])))<<32+ntohl(*((uint32_t*)&packet[20]));
@@ -829,11 +835,18 @@ void rtp_request_resend(seq_t first, uint32_t count, rtsp_conn_info *conn) {
msgsize = sizeof(struct sockaddr_in6);
}
#endif
if (sendto(conn->audio_socket, req, sizeof(req), 0,
(struct sockaddr *)&conn->rtp_client_control_socket, msgsize) == -1) {
char em[1024];
strerror_r(errno, em, sizeof(em));
debug(1, "Error %d using send-to to an audio socket: \"%s\".", errno, em);
uint64_t time_of_sending_fp = get_absolute_time_in_fp();
uint64_t resend_error_backoff_time = (uint64_t)10<<32; // ten seconds
if ((conn->rtp_time_of_last_resend_request_error_fp) || ((time_of_sending_fp-conn->rtp_time_of_last_resend_request_error_fp)>resend_error_backoff_time)) {
if (sendto(conn->audio_socket, req, sizeof(req), 0,
(struct sockaddr *)&conn->rtp_client_control_socket, msgsize) == -1) {
char em[1024];
strerror_r(errno, em, sizeof(em));
debug(1, "Error %d using send-to to an audio socket: \"%s\". ", errno, em);
conn->rtp_time_of_last_resend_request_error_fp = time_of_sending_fp;
} else {
conn->rtp_time_of_last_resend_request_error_fp=0;
}
}
} else {
// if (!request_sent) {