time_of_previous_packet_fp = local_time_now_fp;
}
- if (nread < 0)
- break;
-
- ssize_t plen = nread;
- uint8_t type = packet[1] & ~0x80;
- if (type == 0x60 || type == 0x56) { // audio data / resend
- pktp = packet;
- if (type == 0x56) {
- pktp += 4;
- plen -= 4;
- }
- seq_t seqno = ntohs(*(uint16_t *)(pktp + 2));
- // increment last_seqno and see if it's the same as the incoming seqno
-
- if (type == 0x60) { // regular audio data
- if (last_seqno == -1)
- last_seqno = seqno;
- else {
- last_seqno = (last_seqno + 1) & 0xffff;
- // if (seqno != last_seqno)
- // debug(3, "RTP: Packets out of sequence: expected: %d, got %d.", last_seqno, seqno);
- last_seqno = seqno; // reset warning...
+ if (nread >= 0) {
+ ssize_t plen = nread;
+ uint8_t type = packet[1] & ~0x80;
+ if (type == 0x60 || type == 0x56) { // audio data / resend
+ pktp = packet;
+ if (type == 0x56) {
+ pktp += 4;
+ plen -= 4;
+ }
+ seq_t seqno = ntohs(*(uint16_t *)(pktp + 2));
+ // increment last_seqno and see if it's the same as the incoming seqno
+
+ if (type == 0x60) { // regular audio data
+ if (last_seqno == -1)
+ last_seqno = seqno;
+ else {
+ last_seqno = (last_seqno + 1) & 0xffff;
+ // if (seqno != last_seqno)
+ // debug(3, "RTP: Packets out of sequence: expected: %d, got %d.", last_seqno, seqno);
+ last_seqno = seqno; // reset warning...
+ }
+ } else {
+ debug(1, "Audio Receiver -- Retransmitted Audio Data Packet %u received.", seqno);
}
- } else {
- debug(1, "Audio Port -- Retransmitted Audio Data Packet %u received.", seqno);
- }
- int64_t timestamp = monotonic_timestamp(ntohl(*(uint32_t *)(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.");
+ // if (packet[1]&0x10)
+ // debug(1,"Audio packet Extension bit set.");
- pktp += 12;
- plen -= 12;
+ pktp += 12;
+ plen -= 12;
- // check if packet contains enough content to be reasonable
- if (plen >= 16) {
- if ((config.diagnostic_drop_packet_fraction == 0.0) ||
- (drand48() > config.diagnostic_drop_packet_fraction))
- player_put_packet(seqno, timestamp, pktp, plen, conn);
- else
- debug(3, "Dropping audio packet %u to simulate a bad connection.", seqno);
- continue;
- }
- if (type == 0x56 && seqno == 0) {
- debug(2, "resend-related request packet received, ignoring.");
- continue;
+ // check if packet contains enough content to be reasonable
+ if (plen >= 16) {
+ if ((config.diagnostic_drop_packet_fraction == 0.0) ||
+ (drand48() > config.diagnostic_drop_packet_fraction))
+ player_put_packet(seqno, timestamp, pktp, plen, conn);
+ else
+ debug(3, "Dropping audio packet %u to simulate a bad connection.", seqno);
+ continue;
+ }
+ if (type == 0x56 && seqno == 0) {
+ debug(2, "resend-related request packet received, ignoring.");
+ continue;
+ }
+ debug(1, "Audio receiver -- Unknown RTP packet of type 0x%02X length %d seqno %d", type,
+ nread, seqno);
}
- debug(1, "Audio receiver -- Unknown RTP packet of type 0x%02X length %d seqno %d", type,
- nread, seqno);
+ warn("Audio receiver -- Unknown RTP packet of type 0x%02X length %d.", type, nread);
+ } else {
+ debug(1, "Error receiving an audio packet.");
}
- warn("Audio receiver -- Unknown RTP packet of type 0x%02X length %d.", type, nread);
}
debug(3, "Audio receiver -- Server RTP thread interrupted. terminating.");
if (conn->please_stop != 0) {
break;
}
+
nread = recv(conn->control_socket, packet, sizeof(packet), 0);
// local_time_now = get_absolute_time_in_fp();
// clock_gettime(CLOCK_MONOTONIC,&tn);
// local_time_now=((uint64_t)tn.tv_sec<<32)+((uint64_t)tn.tv_nsec<<32)/1000000000;
- if (nread < 0)
- break;
-
- 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;
+ if (nread >= 0) {
+
+ if ((config.diagnostic_drop_packet_fraction == 0.0) ||
+ (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.
+ // 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.
+ // 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 rtlt = nctohl(&packet[4]); // raw timestamp less latency
+ // uint32_t rt = nctohl(&packet[16]); // raw timestamp
- uint32_t fl = nctohs(&packet[2]); //
+ // 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...
-
- remote_time_of_sync = (uint64_t)nctohl(&packet[8]) << 32;
- remote_time_of_sync += nctohl(&packet[12]);
-
- // debug(1,"Remote Sync Time: %0llx.",remote_time_of_sync);
-
- sync_rtp_timestamp = monotonic_timestamp(nctohl(&packet[16]), conn);
- int64_t rtp_timestamp_less_latency = monotonic_timestamp(nctohl(&packet[4]), conn);
-
- // debug(1,"Sync timestamp is %u.",ntohl(*((uint32_t *)&packet[16])));
+ // 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...
+
+ remote_time_of_sync = (uint64_t)nctohl(&packet[8]) << 32;
+ remote_time_of_sync += nctohl(&packet[12]);
+
+ // debug(1,"Remote Sync Time: %0llx.",remote_time_of_sync);
+
+ sync_rtp_timestamp = monotonic_timestamp(nctohl(&packet[16]), conn);
+ int64_t rtp_timestamp_less_latency = monotonic_timestamp(nctohl(&packet[4]), conn);
+
+ // debug(1,"Sync timestamp is %u.",ntohl(*((uint32_t *)&packet[16])));
+
+ if (config.userSuppliedLatency) {
+ if (config.userSuppliedLatency != conn->latency) {
+ debug(1, "Using the user-supplied latency: %" PRId64 ".",
+ config.userSuppliedLatency);
+ }
+ conn->latency = config.userSuppliedLatency;
+ } else {
+
+ // It seems that the second pair of bytes in the packet indicate whether a fixed
+ // delay of 11,025 frames should be added -- iTunes set this field to 7 and
+ // AirPlay sets it to 4.
+
+ // The value of 11,025 (0.25 seconds) is a guess based on the "Audio-Latency"
+ // parameter
+ // returned by an AE.
+
+ // Sigh, it would be nice to have a published protocol...
+
+ uint16_t flags = nctohs(&packet[2]);
+ int64_t la = sync_rtp_timestamp - rtp_timestamp_less_latency;
+ if (flags == 7)
+ la += config.fixedLatencyOffset;
+ // debug(1,"Latency calculated from the sync packet is %" PRId64 " frames.",la);
+ if ((conn->maximum_latency) && (conn->maximum_latency < la))
+ la = conn->maximum_latency;
+ if ((conn->minimum_latency) && (conn->minimum_latency > la))
+ la = conn->minimum_latency;
+
+ const int max_frames = ((3 * BUFFER_FRAMES * 352) / 4) - 11025;
+
+ if ((la < 0) || (la > max_frames)) {
+ warn("An out-of-range latency request of %" PRId64
+ " frames was ignored. Must be %d frames or less (44,100 frames per second). "
+ "Latency remains at %" PRId64 " frames.",
+ la, max_frames, conn->latency);
+ } else {
+
+ if (la != conn->latency) {
+ conn->latency = la;
+ debug(2, "New latency detected: %" PRId64 ", sync latency: %" PRId64
+ ", minimum latency: %" PRId64 ", maximum "
+ "latency: %" PRId64 ", fixed offset: %" PRId64 ".",
+ la, sync_rtp_timestamp - rtp_timestamp_less_latency, conn->minimum_latency,
+ conn->maximum_latency, config.fixedLatencyOffset);
+ }
+ }
+ }
- if (config.userSuppliedLatency) {
- if (config.userSuppliedLatency != conn->latency) {
- debug(1, "Using the user-supplied latency: %" PRId64 ".", config.userSuppliedLatency);
+ pthread_mutex_lock(&conn->reference_time_mutex);
+
+ // this is for debugging
+ // uint64_t old_remote_reference_time = conn->remote_reference_timestamp_time;
+ // int64_t old_reference_timestamp = conn->reference_timestamp;
+ // int64_t old_latency_delayed_timestamp = conn->latency_delayed_timestamp;
+ conn->remote_reference_timestamp_time = remote_time_of_sync;
+ conn->reference_timestamp_time =
+ remote_time_of_sync - conn->local_to_remote_time_difference;
+ conn->reference_timestamp = sync_rtp_timestamp;
+ conn->latency_delayed_timestamp = rtp_timestamp_less_latency;
+ pthread_mutex_unlock(&conn->reference_time_mutex);
+
+ // this is for debugging
+ /*
+ uint64_t time_difference = remote_time_of_sync - old_remote_reference_time;
+ int64_t reference_frame_difference = sync_rtp_timestamp - old_reference_timestamp;
+ int64_t delayed_frame_difference = rtp_timestamp_less_latency -
+ old_latency_delayed_timestamp;
+
+ if (old_remote_reference_time)
+ debug(1,"Time difference: %" PRIu64 " reference and delayed frame differences: %"
+ PRId64 "
+ and %" PRId64 ", giving rates of %f and %f respectively.",
+ (time_difference*1000000)>>32,reference_frame_difference,delayed_frame_difference,(1.0*(reference_frame_difference*10000000))/((time_difference*10000000)>>32),(1.0*(delayed_frame_difference*10000000))/((time_difference*10000000)>>32));
+ else
+ debug(1,"First sync received");
+ */
+
+ // debug(1,"New Reference timestamp and timestamp time...");
+ // get estimated remote time now
+ // remote_time_now = local_time_now + local_to_remote_time_difference;
+
+ // debug(1,"Sync Time is %lld us late (remote
+ // times).",((remote_time_now-remote_time_of_sync)*1000000)>>32);
+ // debug(1,"Sync Time is %lld us late (local
+ // times).",((local_time_now-reference_timestamp_time)*1000000)>>32);
+ } else {
+ debug(1, "Sync packet received before we got a timing packet back.");
}
- conn->latency = config.userSuppliedLatency;
- } else {
-
- // It seems that the second pair of bytes in the packet indicate whether a fixed
- // delay of 11,025 frames should be added -- iTunes set this field to 7 and
- // AirPlay sets it to 4.
+ } else if (packet[1] == 0xd6) { // resent audio data in the control path -- whaale only?
+ pktp = packet + 4;
+ plen -= 4;
+ seq_t seqno = ntohs(*(uint16_t *)(pktp + 2));
+ debug(1, "Control Receiver -- Retransmitted Audio Data Packet %u received.", seqno);
- // The value of 11,025 (0.25 seconds) is a guess based on the "Audio-Latency" parameter
- // returned by an AE.
+ int64_t timestamp = monotonic_timestamp(ntohl(*(uint32_t *)(pktp + 4)), conn);
- // Sigh, it would be nice to have a published protocol...
+ pktp += 12;
+ plen -= 12;
- uint16_t flags = nctohs(&packet[2]);
- int64_t la = sync_rtp_timestamp - rtp_timestamp_less_latency;
- if (flags == 7)
- la += config.fixedLatencyOffset;
- // debug(1,"Latency calculated from the sync packet is %" PRId64 " frames.",la);
- if ((conn->maximum_latency) && (conn->maximum_latency < la))
- la = conn->maximum_latency;
- if ((conn->minimum_latency) && (conn->minimum_latency > la))
- la = conn->minimum_latency;
-
- const int max_frames = ((3 * BUFFER_FRAMES * 352) / 4) - 11025;
-
- if ((la < 0) || (la > max_frames)) {
- warn("An out-of-range latency request of %" PRId64
- " frames was ignored. Must be %d frames or less (44,100 frames per second). "
- "Latency remains at %" PRId64 " frames.",
- la, max_frames, conn->latency);
+ // check if packet contains enough content to be reasonable
+ if (plen >= 16) {
+ player_put_packet(seqno, timestamp, pktp, plen, conn);
+ continue;
} else {
-
- if (la != conn->latency) {
- conn->latency = la;
- debug(2, "New latency detected: %" PRId64 ", sync latency: %" PRId64
- ", minimum latency: %" PRId64 ", maximum "
- "latency: %" PRId64 ", fixed offset: %" PRId64 ".",
- la, sync_rtp_timestamp - rtp_timestamp_less_latency, conn->minimum_latency,
- conn->maximum_latency, config.fixedLatencyOffset);
- }
+ debug(3, "Too-short retransmitted audio packet received in control port, ignored.");
}
- }
-
- pthread_mutex_lock(&conn->reference_time_mutex);
-
- // this is for debugging
- // uint64_t old_remote_reference_time = conn->remote_reference_timestamp_time;
- // int64_t old_reference_timestamp = conn->reference_timestamp;
- // int64_t old_latency_delayed_timestamp = conn->latency_delayed_timestamp;
- conn->remote_reference_timestamp_time = remote_time_of_sync;
- conn->reference_timestamp_time =
- remote_time_of_sync - conn->local_to_remote_time_difference;
- conn->reference_timestamp = sync_rtp_timestamp;
- conn->latency_delayed_timestamp = rtp_timestamp_less_latency;
- pthread_mutex_unlock(&conn->reference_time_mutex);
-
- // this is for debugging
- /*
- uint64_t time_difference = remote_time_of_sync - old_remote_reference_time;
- int64_t reference_frame_difference = sync_rtp_timestamp - old_reference_timestamp;
- int64_t delayed_frame_difference = rtp_timestamp_less_latency -
- old_latency_delayed_timestamp;
-
- if (old_remote_reference_time)
- debug(1,"Time difference: %" PRIu64 " reference and delayed frame differences: %" PRId64 "
- and %" PRId64 ", giving rates of %f and %f respectively.",
- (time_difference*1000000)>>32,reference_frame_difference,delayed_frame_difference,(1.0*(reference_frame_difference*10000000))/((time_difference*10000000)>>32),(1.0*(delayed_frame_difference*10000000))/((time_difference*10000000)>>32));
- else
- debug(1,"First sync received");
- */
-
- // debug(1,"New Reference timestamp and timestamp time...");
- // get estimated remote time now
- // remote_time_now = local_time_now + local_to_remote_time_difference;
-
- // debug(1,"Sync Time is %lld us late (remote
- // times).",((remote_time_now-remote_time_of_sync)*1000000)>>32);
- // debug(1,"Sync Time is %lld us late (local
- // times).",((local_time_now-reference_timestamp_time)*1000000)>>32);
- } else {
- 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?
- pktp = packet + 4;
- plen -= 4;
- seq_t seqno = ntohs(*(uint16_t *)(pktp + 2));
- debug(1, "Control Port -- Retransmitted Audio Data Packet %u received.", seqno);
-
- int64_t timestamp = monotonic_timestamp(ntohl(*(uint32_t *)(pktp + 4)), conn);
-
- pktp += 12;
- plen -= 12;
-
- // check if packet contains enough content to be reasonable
- if (plen >= 16) {
- player_put_packet(seqno, timestamp, pktp, plen, conn);
- continue;
+ } else
+ debug(1, "Control Receiver -- Unknown RTP packet of type 0x%02X length %d, ignored.",
+ packet[1], nread);
} else {
- debug(3, "Too-short retransmitted audio packet received in control port, ignored.");
+ debug(1, "Control Receiver -- dropping a packet to simulate a bad network.");
}
- } else
- debug(1, "Control Port -- Unknown RTP packet of type 0x%02X length %d, ignored.", packet[1],
- nread);
+ } else {
+ debug(1, "Control Receiver -- error receiving a packet.");
+ }
}
debug(3, "Control RTP thread interrupted. terminating.");
arrival_time = get_absolute_time_in_fp();
// clock_gettime(CLOCK_MONOTONIC,&att);
- if (nread < 0)
- break;
+ if (nread >= 0) {
- // ssize_t plen = nread;
- // debug(1,"Packet Received on Timing Port.");
- if (packet[1] == 0xd3) { // timing reply
- /*
- char obf[4096];
- char *obfp = obf;
- int obfc;
- for (obfc=0;obfc<plen;obfc++) {
- snprintf(obfp, 3, "%02X", packet[obfc]);
- obfp+=2;
- };
- *obfp=0;
- //debug(1,"Timing Packet Received: \"%s\"",obf);
- */
+ // ssize_t plen = nread;
+ // debug(1,"Packet Received on Timing Port.");
+ if (packet[1] == 0xd3) { // timing reply
+ /*
+ char obf[4096];
+ char *obfp = obf;
+ int obfc;
+
for (obfc=0;obfc<plen;obfc++) {
+ snprintf(obfp, 3, "%02X", packet[obfc]);
+ obfp+=2;
+ };
+ *obfp=0;
+ //debug(1,"Timing Packet Received: \"%s\"",obf);
+ */
- // arrival_time = ((uint64_t)att.tv_sec<<32)+((uint64_t)att.tv_nsec<<32)/1000000000;
- // departure_time = ((uint64_t)dtt.tv_sec<<32)+((uint64_t)dtt.tv_nsec<<32)/1000000000;
+ // arrival_time = ((uint64_t)att.tv_sec<<32)+((uint64_t)att.tv_nsec<<32)/1000000000;
+ // departure_time = ((uint64_t)dtt.tv_sec<<32)+((uint64_t)dtt.tv_nsec<<32)/1000000000;
- return_time = arrival_time - conn->departure_time;
+ return_time = arrival_time - conn->departure_time;
- uint64_t rtus = (return_time * 1000000) >> 32;
+ uint64_t rtus = (return_time * 1000000) >> 32;
- if (rtus < 300000) {
+ if (rtus < 300000) {
- // debug(2,"Synchronisation ping return time is %f milliseconds.",(rtus*1.0)/1000);
+ // 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]));
+ // distant_receive_time =
+ // ((uint64_t)ntohl(*((uint32_t*)&packet[16])))<<32+ntohl(*((uint32_t*)&packet[20]));
- distant_receive_time = (uint64_t)nctohl(&packet[16]) << 32;
- distant_receive_time += nctohl(&packet[20]);
+ distant_receive_time = (uint64_t)nctohl(&packet[16]) << 32;
+ distant_receive_time += nctohl(&packet[20]);
- // distant_transmit_time =
- // ((uint64_t)ntohl(*((uint32_t*)&packet[24])))<<32+ntohl(*((uint32_t*)&packet[28]));
+ // distant_transmit_time =
+ // ((uint64_t)ntohl(*((uint32_t*)&packet[24])))<<32+ntohl(*((uint32_t*)&packet[28]));
- distant_transmit_time = (uint64_t)nctohl(&packet[24]) << 32;
- distant_transmit_time += nctohl(&packet[28]);
+ distant_transmit_time = (uint64_t)nctohl(&packet[24]) << 32;
+ distant_transmit_time += nctohl(&packet[28]);
- // processing_time = distant_transmit_time - distant_receive_time;
+ // processing_time = distant_transmit_time - distant_receive_time;
- // debug(1,"Return trip time: %lluuS, remote processing time:
- // %lluuS.",(return_time*1000000)>>32,(processing_time*1000000)>>32);
+ // debug(1,"Return trip time: %lluuS, remote processing time:
+ // %lluuS.",(return_time*1000000)>>32,(processing_time*1000000)>>32);
- uint64_t local_time_by_remote_clock = distant_transmit_time + return_time / 2;
+ uint64_t local_time_by_remote_clock = distant_transmit_time + return_time / 2;
- unsigned int cc;
- for (cc = time_ping_history - 1; cc > 0; cc--) {
- conn->time_pings[cc] = conn->time_pings[cc - 1];
- conn->time_pings[cc].dispersion =
- (conn->time_pings[cc].dispersion * 110) /
- 100; // make the dispersions 'age' by this rational factor
- }
- // these are for diagnostics only -- not used
- conn->time_pings[0].local_time = arrival_time;
- conn->time_pings[0].remote_time = distant_transmit_time;
-
- conn->time_pings[0].local_to_remote_difference = local_time_by_remote_clock - arrival_time;
- conn->time_pings[0].dispersion = return_time;
- if (conn->time_ping_count < time_ping_history)
- conn->time_ping_count++;
-
- uint64_t local_time_chosen = arrival_time;
- ;
- // uint64_t remote_time_chosen = distant_transmit_time;
- // now pick the timestamp with the lowest dispersion
- uint64_t l2rtd = conn->time_pings[0].local_to_remote_difference;
- uint64_t tld = conn->time_pings[0].dispersion;
- // chosen = 0;
- for (cc = 1; cc < conn->time_ping_count; cc++)
- if (conn->time_pings[cc].dispersion < tld) {
- l2rtd = conn->time_pings[cc].local_to_remote_difference;
- // chosen = cc;
- tld = conn->time_pings[cc].dispersion;
- local_time_chosen = conn->time_pings[cc].local_time;
- // remote_time_chosen = conn->time_pings[cc].remote_time;
+ unsigned int cc;
+ for (cc = time_ping_history - 1; cc > 0; cc--) {
+ conn->time_pings[cc] = conn->time_pings[cc - 1];
+ conn->time_pings[cc].dispersion =
+ (conn->time_pings[cc].dispersion * 110) /
+ 100; // make the dispersions 'age' by this rational factor
+ }
+ // these are for diagnostics only -- not used
+ conn->time_pings[0].local_time = arrival_time;
+ conn->time_pings[0].remote_time = distant_transmit_time;
+
+ conn->time_pings[0].local_to_remote_difference =
+ local_time_by_remote_clock - arrival_time;
+ conn->time_pings[0].dispersion = return_time;
+ if (conn->time_ping_count < time_ping_history)
+ conn->time_ping_count++;
+
+ uint64_t local_time_chosen = arrival_time;
+
+ // uint64_t remote_time_chosen = distant_transmit_time;
+ // now pick the timestamp with the lowest dispersion
+ uint64_t l2rtd = conn->time_pings[0].local_to_remote_difference;
+ uint64_t tld = conn->time_pings[0].dispersion;
+ // chosen = 0;
+ for (cc = 1; cc < conn->time_ping_count; cc++)
+ if (conn->time_pings[cc].dispersion < tld) {
+ l2rtd = conn->time_pings[cc].local_to_remote_difference;
+ // chosen = cc;
+ tld = conn->time_pings[cc].dispersion;
+ local_time_chosen = conn->time_pings[cc].local_time;
+ // remote_time_chosen = conn->time_pings[cc].remote_time;
+ }
+ // int64_t ji;
+
+ if (conn->time_ping_count > 1) {
+ if (l2rtd > conn->local_to_remote_time_difference) {
+ local_to_remote_time_jitters =
+ local_to_remote_time_jitters + l2rtd - conn->local_to_remote_time_difference;
+ // ji = l2rtd - conn->local_to_remote_time_difference;
+ } else {
+ local_to_remote_time_jitters =
+ local_to_remote_time_jitters + conn->local_to_remote_time_difference - l2rtd;
+ // ji = -(conn->local_to_remote_time_difference - l2rtd);
+ }
+ local_to_remote_time_jitters_count += 1;
+ }
+ // uncomment below to print jitter between client's clock and oour clock
+ // int64_t rtus = (tld*1000000)>>32; ji = (ji*1000000)>>32; debug(1,"Choosing time
+ // difference
+ // with dispersion of %lld us with delta of %lld us",rtus,ji);
+
+ conn->local_to_remote_time_difference = l2rtd;
+ if (first_local_to_remote_time_difference == 0) {
+ first_local_to_remote_time_difference = conn->local_to_remote_time_difference;
+ // first_local_to_remote_time_difference_time = get_absolute_time_in_fp();
}
- // int64_t ji;
- if (conn->time_ping_count > 1) {
- if (l2rtd > conn->local_to_remote_time_difference) {
- local_to_remote_time_jitters =
- local_to_remote_time_jitters + l2rtd - conn->local_to_remote_time_difference;
- // ji = l2rtd - conn->local_to_remote_time_difference;
+ // int64_t clock_drift;
+ // int64_t clock_drift_in_usec;
+ // double clock_drift_ppm = 0.0;
+ if (first_local_time == 0) {
+ first_local_time = local_time_chosen;
+ // first_remote_time = remote_time_chosen;
+ // clock_drift = 0;
} else {
- local_to_remote_time_jitters =
- local_to_remote_time_jitters + conn->local_to_remote_time_difference - l2rtd;
- // ji = -(conn->local_to_remote_time_difference - l2rtd);
+ // uint64_t local_time_change = local_time_chosen - first_local_time;
+ // uint64_t remote_time_change = remote_time_chosen - first_remote_time;
+
+ /*
+ if (remote_time_change >= local_time_change)
+ clock_drift = remote_time_change - local_time_change;
+ else
+ clock_drift = -(local_time_change - remote_time_change);
+ */
+ /*
+ if (clock_drift >= 0)
+ clock_drift_in_usec = (clock_drift * 1000000) >> 32;
+ else
+ clock_drift_in_usec = -(((-clock_drift) * 1000000) >> 32);
+ */
+
+ // clock_drift_ppm = (1.0 * clock_drift_in_usec) / (local_time_change >> 32);
}
- local_to_remote_time_jitters_count += 1;
- }
- // uncomment below to print jitter between client's clock and oour clock
- // int64_t rtus = (tld*1000000)>>32; ji = (ji*1000000)>>32; debug(1,"Choosing time
- // difference
- // with dispersion of %lld us with delta of %lld us",rtus,ji);
-
- conn->local_to_remote_time_difference = l2rtd;
- if (first_local_to_remote_time_difference == 0) {
- first_local_to_remote_time_difference = conn->local_to_remote_time_difference;
- // first_local_to_remote_time_difference_time = get_absolute_time_in_fp();
- }
- // int64_t clock_drift;
- // int64_t clock_drift_in_usec;
- // double clock_drift_ppm = 0.0;
- if (first_local_time == 0) {
- first_local_time = local_time_chosen;
- // first_remote_time = remote_time_chosen;
- // clock_drift = 0;
+ int64_t source_drift_usec;
+ if (conn->play_segment_reference_frame != 0) {
+ int64_t reference_timestamp;
+ uint64_t reference_timestamp_time, remote_reference_timestamp_time;
+ get_reference_timestamp_stuff(&reference_timestamp, &reference_timestamp_time,
+ &remote_reference_timestamp_time, conn);
+ uint64_t frame_difference = 0;
+ if (reference_timestamp >= conn->play_segment_reference_frame)
+ frame_difference =
+ (uint64_t)reference_timestamp - (uint64_t)conn->play_segment_reference_frame;
+ else // rollover
+ frame_difference = (uint64_t)reference_timestamp + 0x100000000 -
+ (uint64_t)conn->play_segment_reference_frame;
+ uint64_t frame_time_difference_calculated =
+ (((uint64_t)frame_difference << 32) / 44100);
+ uint64_t frame_time_difference_actual =
+ remote_reference_timestamp_time -
+ conn->play_segment_reference_frame_remote_time; // this is all done by reference to
+ // the
+ // sources' system clock
+ // debug(1,"%llu frames since play started, %llu usec calculated, %llu usec
+ // actual",frame_difference, (frame_time_difference_calculated*1000000)>>32,
+ // (frame_time_difference_actual*1000000)>>32);
+ if (frame_time_difference_calculated >=
+ frame_time_difference_actual) // i.e. if the time it should have taken to send the
+ // packets is greater than the actual time difference
+ // measured on the source clock
+ // then the source DAC's clock is running fast relative to the source system clock
+ source_drift_usec = frame_time_difference_calculated - frame_time_difference_actual;
+ else
+ // otherwise the source DAC's clock is running slow relative to the source system
+ // clock
+ source_drift_usec =
+ -(frame_time_difference_actual - frame_time_difference_calculated);
+ } else
+ source_drift_usec = 0;
+ source_drift_usec = (source_drift_usec * 1000000) >> 32; // turn it to microseconds
+
+ // long current_delay = 0;
+ // if (config.output->delay) {
+ // config.output->delay(¤t_delay);
+ //}
+ // Useful for troubleshooting:
+ // debug(1, "clock_drift_ppm %f\tchosen %5d\tsource_drift_usec
+ // %10.1lld\treturn_time_in_usec
+ // %10.1llu",
+ // clock_drift_ppm,
+ // chosen,
+ //(session_corrections*1000000)/44100,
+ // current_delay,
+ // source_drift_usec,
+ // buffer_occupancy,
+ //(return_time*1000000)>>32);
} else {
- // uint64_t local_time_change = local_time_chosen - first_local_time;
- // uint64_t remote_time_change = remote_time_chosen - first_remote_time;
-
- /*
- if (remote_time_change >= local_time_change)
- clock_drift = remote_time_change - local_time_change;
- else
- clock_drift = -(local_time_change - remote_time_change);
- */
- /*
- if (clock_drift >= 0)
- clock_drift_in_usec = (clock_drift * 1000000) >> 32;
- else
- clock_drift_in_usec = -(((-clock_drift) * 1000000) >> 32);
- */
-
- // clock_drift_ppm = (1.0 * clock_drift_in_usec) / (local_time_change >> 32);
+ debug(1, "Time ping turnaround time: %lld us -- it looks like a timing ping was lost.",
+ rtus);
}
-
- int64_t source_drift_usec;
- if (conn->play_segment_reference_frame != 0) {
- int64_t reference_timestamp;
- uint64_t reference_timestamp_time, remote_reference_timestamp_time;
- get_reference_timestamp_stuff(&reference_timestamp, &reference_timestamp_time,
- &remote_reference_timestamp_time, conn);
- uint64_t frame_difference = 0;
- if (reference_timestamp >= conn->play_segment_reference_frame)
- frame_difference =
- (uint64_t)reference_timestamp - (uint64_t)conn->play_segment_reference_frame;
- else // rollover
- frame_difference = (uint64_t)reference_timestamp + 0x100000000 -
- (uint64_t)conn->play_segment_reference_frame;
- uint64_t frame_time_difference_calculated = (((uint64_t)frame_difference << 32) / 44100);
- uint64_t frame_time_difference_actual =
- remote_reference_timestamp_time -
- conn->play_segment_reference_frame_remote_time; // this is all done by reference to
- // the
- // sources' system clock
- // debug(1,"%llu frames since play started, %llu usec calculated, %llu usec
- // actual",frame_difference, (frame_time_difference_calculated*1000000)>>32,
- // (frame_time_difference_actual*1000000)>>32);
- if (frame_time_difference_calculated >=
- frame_time_difference_actual) // i.e. if the time it should have taken to send the
- // packets is greater than the actual time difference
- // measured on the source clock
- // then the source DAC's clock is running fast relative to the source system clock
- source_drift_usec = frame_time_difference_calculated - frame_time_difference_actual;
- else
- // otherwise the source DAC's clock is running slow relative to the source system clock
- source_drift_usec = -(frame_time_difference_actual - frame_time_difference_calculated);
- } else
- source_drift_usec = 0;
- source_drift_usec = (source_drift_usec * 1000000) >> 32; // turn it to microseconds
-
- // long current_delay = 0;
- // if (config.output->delay) {
- // config.output->delay(¤t_delay);
- //}
- // Useful for troubleshooting:
- // debug(1, "clock_drift_ppm %f\tchosen %5d\tsource_drift_usec %10.1lld\treturn_time_in_usec
- // %10.1llu",
- // clock_drift_ppm,
- // chosen,
- //(session_corrections*1000000)/44100,
- // current_delay,
- // source_drift_usec,
- // buffer_occupancy,
- //(return_time*1000000)>>32);
} else {
- debug(1, "Time ping turnaround time: %lld us -- it looks like a timing ping was lost.",
- rtus);
+ debug(1, "Timing port -- Unknown RTP packet of type 0x%02X length %d.", packet[1], nread);
}
} else {
- debug(1, "Timing port -- Unknown RTP packet of type 0x%02X length %d.", packet[1], nread);
+ debug(1, "Timing receiver -- error receiving a packet.");
}
}
void rtp_request_resend(seq_t first, uint32_t count, rtsp_conn_info *conn) {
if (conn->rtp_running) {
// if (!request_sent) {
- debug(1, "requesting resend of %d packets starting at %u.", count, first);
+ debug(2, "requesting resend of %d packets starting at %u.", count, first);
// request_sent = 1;
//}
projects / thirdparty / shairport-sync.git / commitdiff
